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authorMarcin Juszkiewicz <hrw@openembedded.org>2006-11-27 22:05:13 +0000
committerMarcin Juszkiewicz <hrw@openembedded.org>2006-11-27 22:05:13 +0000
commit6ee18ab0f2901d2d87a94e761f29d088c6e5a204 (patch)
treed0ad6ed2083db46adda2b0d7dac3bf8671163dd8 /packages
parent2a03b41eb0a2aeb9e556bc088b002a1ad51f0f9e (diff)
parent313c82e3bd14e15219a0ce799e3862d64405f25b (diff)
downloadopenembedded-6ee18ab0f2901d2d87a94e761f29d088c6e5a204.tar.gz
openembedded-6ee18ab0f2901d2d87a94e761f29d088c6e5a204.tar.bz2
openembedded-6ee18ab0f2901d2d87a94e761f29d088c6e5a204.zip
merge of '755167f16f9b0fce49c13343f86891cd973fc1cb'
and 'aaef2980a4428dedfb31b38d6894621134b0faa8'
Diffstat (limited to 'packages')
-rwxr-xr-xpackages/initscripts/initscripts-1.0/checkroot.sh14
-rw-r--r--packages/initscripts/initscripts_1.0.bb2
-rw-r--r--packages/linux/linux-openzaurus-2.6.17/asoc-v0.12.4_2.6.17.patch31713
-rw-r--r--packages/linux/linux-openzaurus_2.6.17.bb7
-rw-r--r--packages/mplayer/files/imageon-video_out.patch20
-rw-r--r--packages/mplayer/mplayer_1.0pre8.bb5
-rw-r--r--packages/ossie/ossie-demo_svn.bb1
-rw-r--r--packages/ossie/ossie-interpolator_svn.bb20
-rw-r--r--packages/ossie/ossie-modulator_svn.bb20
-rw-r--r--packages/ossie/ossie-randombits_svn.bb20
-rw-r--r--packages/ossie/ossie-sigproc_svn.bb19
-rw-r--r--packages/ossie/ossie-tx-random-data_svn.bb20
-rw-r--r--packages/tasks/task-ossie.bb5
13 files changed, 31853 insertions, 13 deletions
diff --git a/packages/initscripts/initscripts-1.0/checkroot.sh b/packages/initscripts/initscripts-1.0/checkroot.sh
index df3035371b..44db23707e 100755
--- a/packages/initscripts/initscripts-1.0/checkroot.sh
+++ b/packages/initscripts/initscripts-1.0/checkroot.sh
@@ -148,7 +148,10 @@ else
# 2 or larger. A return code of 1 indicates that filesystem
# errors were corrected but that the boot may proceed.
#
- if test "$?" -gt 1
+
+ echo "RETURNCODE: [$RTC]"
+
+ if test "$RTC" -gt 3
then
# Since this script is run very early in the boot-process, it should be safe to assume that the
@@ -159,13 +162,14 @@ else
# Surprise! Re-directing from a HERE document (as in
# "cat << EOF") won't work, because the root is read-only.
echo
- echo "fsck failed. Please repair manually and reboot. Please note"
- echo "that the root filesystem is currently mounted read-only. To"
- echo "remount it read-write:"
+ echo "fsck failed. Please repair manually and reboot. "
+ echo "Please note that the root filesystem is currently "
+ echo "mounted read-only. To remount it read-write:"
echo
echo " # mount -n -o remount,rw /"
echo
- echo "CONTROL-D will exit from this shell and REBOOT the system."
+ echo "CONTROL-D will exit from this shell"
+ echo "and REBOOT the system."
echo
# Start a single user shell on the console
/sbin/sulogin $CONSOLE
diff --git a/packages/initscripts/initscripts_1.0.bb b/packages/initscripts/initscripts_1.0.bb
index 306629732a..39662eb31a 100644
--- a/packages/initscripts/initscripts_1.0.bb
+++ b/packages/initscripts/initscripts_1.0.bb
@@ -5,7 +5,7 @@ DEPENDS = "makedevs"
DEPENDS_openzaurus = "makedevs virtual/kernel"
RDEPENDS = "makedevs"
LICENSE = "GPL"
-PR = "r82"
+PR = "r83"
SRC_URI = "file://halt \
file://ramdisk \
diff --git a/packages/linux/linux-openzaurus-2.6.17/asoc-v0.12.4_2.6.17.patch b/packages/linux/linux-openzaurus-2.6.17/asoc-v0.12.4_2.6.17.patch
new file mode 100644
index 0000000000..7fa3822bba
--- /dev/null
+++ b/packages/linux/linux-openzaurus-2.6.17/asoc-v0.12.4_2.6.17.patch
@@ -0,0 +1,31713 @@
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/DAI.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/DAI.txt
+@@ -0,0 +1,546 @@
++ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
++SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM.
++
++
++AC97
++====
++
++ AC97 is a five wire interface commonly found on many PC sound cards. It is
++now also popular in many portable devices. This DAI has a reset line and time
++multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
++The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
++frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
++frame is 21uS long and is divided into 13 time slots.
++
++The AC97 specification can be found at :-
++http://www.intel.com/design/chipsets/audio/ac97_r23.pdf
++
++
++I2S
++===
++
++ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
++Rx lines are used for audio transmision, whilst the bit clock (BCLK) and
++left/right clock (LRC) synchronise the link. I2S is flexible in that either the
++controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
++usually varies depending on the sample rate and the master system clock
++(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
++ADC and DAC LRCLK's, this allows for similtanious capture and playback at
++different sample rates.
++
++I2S has several different operating modes:-
++
++ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
++ transition.
++
++ o Left Justified - MSB is transmitted on transition of LRC.
++
++ o Right Justified - MSB is transmitted sample size BCLK's before LRC
++ transition.
++
++PCM
++===
++
++PCM is another 4 wire interface, very similar to I2S, that can support a more
++flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
++to synchronise the link whilst the Tx and Rx lines are used to transmit and
++receive the audio data. Bit clock usually varies depending on sample rate
++whilst sync runs at the sample rate. PCM also supports Time Division
++Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This
++is sometimes referred to as network mode).
++
++Common PCM operating modes:-
++
++ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
++
++ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
++
++
++ASoC DAI Configuration
++======================
++
++Every CODEC DAI and SoC DAI must have their capabilities defined in order to
++be configured together at runtime when the audio and clocking parameters are
++known. This is achieved by creating an array of struct snd_soc_hw_mode in the
++the CODEC and SoC interface drivers. Each element in the array describes a DAI
++mode and each mode is usually based upon the DAI system clock to sample rate
++ratio (FS).
++
++i.e. 48k sample rate @ 256 FS = sytem clock of 12.288 MHz
++ 48000 * 256 = 12288000
++
++The CPU and Codec DAI modes are then ANDed together at runtime to determine the
++rutime DAI configuration for both the Codec and CPU.
++
++When creating a new codec or SoC DAI it's probably best to start of with a few
++sample rates first and then test your interface.
++
++struct snd_soc_dai_mode is defined (in soc.h) as:-
++
++/* SoC DAI mode */
++struct snd_soc_dai_mode {
++ u16 fmt; /* SND_SOC_DAIFMT_* */
++ u16 tdm; /* SND_SOC_HWTDM_* */
++ u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */
++ u16 pcmrate; /* SND_SOC_HWRATE_* */
++ u16 pcmdir:2; /* SND_SOC_HWDIR_* */
++ u16 flags:8; /* hw flags */
++ u16 fs; /* mclk to rate divider */
++ u64 bfs; /* mclk to bclk dividers */
++ unsigned long priv; /* private mode data */
++};
++
++fmt:
++----
++This field defines the DAI mode hardware format (e.g. I2S settings) and
++supports the following settings:-
++
++ 1) hardware DAI formats
++
++#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */
++#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */
++#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */
++#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM */
++#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM */
++#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */
++
++ 2) hw DAI signal inversions
++
++#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */
++#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */
++#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */
++#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */
++
++ 3) hw clock masters
++ This is wrt the codec, the inverse is true for the interface
++ i.e. if the codec is clk and frm master then the interface is
++ clk and frame slave.
++
++#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */
++#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */
++#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */
++#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */
++
++At least one option from each section must be selected. Multiple selections are
++also supported e.g.
++
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
++ SND_SOC_DAIFMT_IB_IF
++
++
++tdm:
++------
++This field defines the Time Division Multiplexing left and right word
++positions for the DAI mode if applicable. Set to SND_SOC_DAITDM_LRDW(0,0) for
++no TDM.
++
++
++pcmfmt:
++---------
++The hardware PCM format. This describes the PCM formats supported by the DAI
++mode e.g.
++
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
++ SNDRV_PCM_FORMAT_S24_3LE
++
++pcmrate:
++----------
++The PCM sample rates supported by the DAI mode. e.g.
++
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000
++
++
++pcmdir:
++---------
++The stream directions supported by this mode. e.g. playback and capture
++
++
++flags:
++--------
++The DAI hardware flags supported by the mode.
++
++/* use bfs mclk divider mode (BCLK = MCLK / x) */
++#define SND_SOC_DAI_BFS_DIV 0x1
++/* use bfs rate mulitplier (BCLK = RATE * x)*/
++#define SND_SOC_DAI_BFS_RATE 0x2
++/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */
++#define SND_SOC_DAI_BFS_RCW 0x4
++/* capture and playback can use different clocks */
++#define SND_SOC_DAI_ASYNC 0x8
++
++NOTE: Bitclock division and mulitiplication modes can be safely matched by the
++core logic.
++
++
++fs:
++-----
++The FS supported by this DAI mode FS is the ratio between the system clock and
++the sample rate. See above
++
++bfs:
++------
++BFS is the ratio of BCLK to MCLK or the ratio of BCLK to sample rate (this
++depends on the codec or CPU DAI).
++
++The BFS supported by the DAI mode. This can either be the ratio between the
++bitclock (BCLK) and the sample rate OR the ratio between the system clock and
++the sample rate. Depends on the flags above.
++
++priv:
++-----
++private codec mode data.
++
++
++
++Examples
++========
++
++Note that Codec DAI and CPU DAI examples are interchangeable in these examples
++as long as the bus master is reversed. i.e.
++
++ SND_SOC_DAIFMT_CBM_CFM would become SND_SOC_DAIFMT_CBS_CFS
++ and vice versa.
++
++This applies to all SND_SOC_DAIFMT_CB*_CF*.
++
++Example 1
++---------
++
++Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
++BCLK of either MCLK/2 or MCLK/4.
++
++ /* codec master */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(2) | SND_SOC_FSBD(4),
++ }
++
++
++Example 2
++---------
++Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
++BCLK of either Rate * 32 or Rate * 64.
++
++ /* codec master */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 32,
++ },
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++
++
++Example 3
++---------
++Codec that runs at 8k & 48k @ 256FS in master mode, can generate a BCLK that
++is a multiple of Rate * channels * word size. (RCW) i.e.
++
++ BCLK = 8000 * 2 * 16 (8k, stereo, 16bit)
++ = 256kHz
++
++This codecs supports a RCW multiple of 1,2
++
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1) | SND_SOC_FSBW(2),
++ }
++
++
++Example 4
++---------
++Codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
++BCLK of either Rate * 32 or Rate * 64. Codec can also run in slave mode as long
++as BCLK is rate * 32 or rate * 64.
++
++ /* codec master */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 32,
++ },
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++
++ /* codec slave */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = 32,
++ },
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = 64,
++ },
++
++
++Example 5
++---------
++Codec that only runs at 8k, 16k, 32k, 48k, 96k @ 128FS, 192FS & 256FS in master
++mode and can generate a BCLK of MCLK / (1,2,4,8,16). Codec can also run in slave
++mode as and does not care about FS or BCLK (as long as there is enough bandwidth).
++
++ #define CODEC_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++ #define CODEC_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
++
++ /* codec master @ 128, 192 & 256 FS */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = CODEC_RATES,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 128,
++ .bfs = CODEC_FSB,
++ },
++
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = CODEC_RATES,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 192,
++ .bfs = CODEC_FSB
++ },
++
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = CODEC_RATES,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = CODEC_FSB,
++ },
++
++ /* codec slave */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = CODEC_RATES,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++
++
++Example 6
++---------
++Codec that only runs at 8k, 44.1k, 48k @ different FS in master mode (for use
++with a fixed MCLK) and can generate a BCLK of MCLK / (1,2,4,8,16).
++Codec can also run in slave mode as and does not care about FS or BCLK (as long
++as there is enough bandwidth). Codec can support 16, 24 and 32 bit PCM sample
++sizes.
++
++ #define CODEC_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++ #define CODEC_PCM_FORMATS \
++ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
++ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | SNDRV_PCM_FORMAT_S32_LE)
++
++ /* codec master */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = CODEC_FSB,
++ },
++
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 272,
++ .bfs = CODEC_FSB,
++ },
++
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = CODEC_FSB,
++ },
++
++ /* codec slave */
++ {
++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
++ .pcmrate = CODEC_RATES,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++
++
++Example 7
++---------
++AC97 Codec that does not support VRA (i.e only runs at 48k).
++
++ #define AC97_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++ #define AC97_PCM_FORMATS \
++ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S18_3LE | \
++ SNDRV_PCM_FORMAT_S20_3LE)
++
++ /* AC97 with no VRA */
++ {
++ .pcmfmt = AC97_PCM_FORMATS,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ }
++
++
++Example 8
++---------
++
++CPU DAI that supports 8k - 48k @ 256FS and BCLK = MCLK / 4 in master mode.
++Slave mode (CPU DAI is FRAME master) supports 8k - 96k at any FS as long as
++BCLK = 64 * rate. (Intel XScale I2S controller).
++
++ #define PXA_I2S_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF)
++
++ #define PXA_I2S_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++ #define PXA_I2S_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++ /* priv is divider */
++ static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = {
++ /* pxa2xx I2S frame and clock master modes */
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x48,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x34,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x24,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x1a,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0xd,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0xc,
++ },
++
++ /* pxa2xx I2S frame master and clock slave mode */
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = PXA_I2S_RATES,
++ .pcmdir = PXA_I2S_DIR,
++ .fs = SND_SOC_FS_ALL,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .bfs = 64,
++ .priv = 0x48,
++ },
++};
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/clocking.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/clocking.txt
+@@ -0,0 +1,314 @@
++Audio Clocking
++==============
++
++This text describes the audio clocking terms in ASoC and digital audio in
++general. Note: Audio clocking can be complex !
++
++
++Master Clock
++------------
++
++Every audio subsystem is driven by a master clock (sometimes refered to as MCLK
++or SYSCLK). This audio master clock can be derived from a number of sources
++(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
++audio playback and capture sample rates.
++
++Some master clocks (e.g. PLL's and CPU based clocks) are configuarble in that
++their speed can be altered by software (depending on the system use and to save
++power). Other master clocks are fixed at at set frequency (i.e. crystals).
++
++
++DAI Clocks
++----------
++The Digital Audio Interface is usually driven by a Bit Clock (often referred to
++as BCLK). This clock is used to drive the digital audio data across the link
++between the codec and CPU.
++
++The DAI also has a frame clock to signal the start of each audio frame. This
++clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
++runs at exactly the sample rate (LRC = Rate).
++
++Bit Clock can be generated as follows:-
++
++BCLK = MCLK / x
++
++ or
++
++BCLK = LRC * x
++
++ or
++
++BCLK = LRC * Channels * Word Size
++
++This relationship depends on the codec or SoC CPU in particular. ASoC can quite
++easily match BCLK generated by division (SND_SOC_DAI_BFS_DIV) with BCLK by
++multiplication (SND_SOC_DAI_BFS_RATE) or BCLK generated by
++Rate * Channels * Word size (RCW or SND_SOC_DAI_BFS_RCW).
++
++
++ASoC Clocking
++-------------
++
++The ASoC core determines the clocking for each particular configuration at
++runtime. This is to allow for dynamic audio clocking wereby the audio clock is
++variable and depends on the system state or device usage scenario. i.e. a voice
++call requires slower clocks (and hence less power) than MP3 playback.
++
++ASoC will call the config_sysclock() function for the target machine during the
++audio parameters configuration. The function is responsible for then clocking
++the machine audio subsytem and returning the audio clock speed to the core.
++This function should also call the codec and cpu DAI clock_config() functions
++to configure their respective internal clocking if required.
++
++
++ASoC Clocking Control Flow
++--------------------------
++
++The ASoC core will call the machine drivers config_sysclock() when most of the
++DAI capabilities are known. The machine driver is then responsible for calling
++the codec and/or CPU DAI drivers with the selected capabilities and the current
++MCLK. Note that the machine driver is also resonsible for setting the MCLK (and
++enabling it).
++
++ (1) Match Codec and CPU DAI capabilities. At this point we have
++ matched the majority of the DAI fields and now need to make sure this
++ mode is currently clockable.
++
++ (2) machine->config_sysclk() is now called with the matched DAI FS, sample
++ rate and BCLK master. This function then gets/sets the current audio
++ clock (depening on usage) and calls the codec and CPUI DAI drivers with
++ the FS, rate, BCLK master and MCLK.
++
++ (3) Codec/CPU DAI config_sysclock(). This function checks that the FS, rate,
++ BCLK master and MCLK are acceptable for the codec or CPU DAI. It also
++ sets the DAI internal state to work with said clocks.
++
++The config_sysclk() functions for CPU, codec and machine should return the MCLK
++on success and 0 on failure.
++
++
++Examples (b = BCLK, l = LRC)
++============================
++
++Example 1
++---------
++
++Simple codec that only runs at 48k @ 256FS in master mode.
++
++CPU only runs as slave DAI, however it generates a variable MCLK.
++
++ -------- ---------
++ | | <----mclk--- | |
++ | Codec |b -----------> | CPU |
++ | |l -----------> | |
++ | | | |
++ -------- ---------
++
++The codec driver has the following config_sysclock()
++
++ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++ {
++ /* make sure clock is 256 * rate */
++ if(info->rate << 8 == clk) {
++ dai->mclk = clk;
++ return clk;
++ }
++
++ return 0;
++ }
++
++The CPU I2S DAI driver has the following config_sysclk()
++
++ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++ {
++ /* can we support this clk */
++ if(set_audio_clk(clk) < 0)
++ return -EINVAL;
++
++ dai->mclk = clk;
++ return dai->clk;
++ }
++
++The machine driver config_sysclk() in this example is as follows:-
++
++ unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++ {
++ int clk = info->rate * info->fs;
++
++ /* check that CPU can deliver clock */
++ if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0)
++ return -EINVAL;
++
++ /* can codec work with this clock */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk);
++ }
++
++
++Example 2
++---------
++
++Codec that can master at 8k and 48k at various FS (and hence supports a fixed
++set of input MCLK's) and can also be slave at various FS .
++
++The CPU can master at 8k and 48k @256 FS and can be slave at any FS.
++
++MCLK is a 12.288MHz crystal on this machine.
++
++ -------- ---------
++ | | <---xtal---> | |
++ | Codec |b <----------> | CPU |
++ | |l <----------> | |
++ | | | |
++ -------- ---------
++
++
++The codec driver has the following config_sysclock()
++
++ /* supported input clocks */
++ const static int hifi_clks[] = {11289600, 12000000, 12288000,
++ 16934400, 18432000};
++
++ static unsigned int config_hsysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++ {
++ int i;
++
++ /* is clk supported */
++ for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) {
++ if(clk == hifi_clks[i]) {
++ dai->mclk = clk;
++ return clk;
++ }
++ }
++
++ /* this clk is not supported */
++ return 0;
++ }
++
++The CPU I2S DAI driver has the following config_sysclk()
++
++ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++ {
++ /* are we master or slave */
++ if (info->bclk_master &
++ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
++
++ /* we can only master @ 256FS */
++ if(info->rate << 8 == clk) {
++ dai->mclk = clk;
++ return dai->mclk;
++ }
++ } else {
++ /* slave we can run at any FS */
++ dai->mclk = clk;
++ return dai->mclk;
++ }
++
++ /* not supported */
++ return dai->clk;
++ }
++
++The machine driver config_sysclk() in this example is as follows:-
++
++ unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++ {
++ int clk = 12288000; /* 12.288MHz */
++
++ /* who's driving the link */
++ if (info->bclk_master &
++ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
++ /* codec master */
++
++ /* check that CPU can work with clock */
++ if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0)
++ return -EINVAL;
++
++ /* can codec work with this clock */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk);
++ } else {
++ /* cpu master */
++
++ /* check that codec can work with clock */
++ if(rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk) < 0)
++ return -EINVAL;
++
++ /* can CPU work with this clock */
++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk);
++ }
++ }
++
++
++
++Example 3
++---------
++
++Codec that masters at 8k ... 48k @256 FS. Codec can also be slave and
++doesn't care about FS. The codec has an internal PLL and dividers to generate
++the necessary internal clocks (for 256FS).
++
++CPU can only be slave and doesn't care about FS.
++
++MCLK is a non controllable 13MHz clock from the CPU.
++
++
++ -------- ---------
++ | | <----mclk--- | |
++ | Codec |b <----------> | CPU |
++ | |l <----------> | |
++ | | | |
++ -------- ---------
++
++The codec driver has the following config_sysclock()
++
++ /* valid PCM clock dividers * 2 */
++ static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16};
++
++ static unsigned int config_vsysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++ {
++ int i, j, best_clk = info->fs * info->rate;
++
++ /* can we run at this clk without the PLL ? */
++ for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) {
++ if ((best_clk >> 1) * pcm_divs[i] == clk) {
++ dai->pll_in = 0;
++ dai->clk_div = pcm_divs[i];
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++
++ /* now check for PLL support */
++ for (i = 0; i < ARRAY_SIZE(pll_div); i++) {
++ if (pll_div[i].pll_in == clk) {
++ for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) {
++ if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) {
++ dai->pll_in = clk;
++ dai->pll_out = pll_div[i].pll_out;
++ dai->clk_div = pcm_divs[j];
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++ }
++ }
++
++ /* this clk is not supported */
++ return 0;
++ }
++
++
++The CPU I2S DAI driver has the does not need a config_sysclk() as it can slave
++at any FS.
++
++ unsigned int config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++ {
++ /* codec has pll that generates mclk from 13MHz xtal */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000);
++ }
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/codec.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/codec.txt
+@@ -0,0 +1,232 @@
++ASoC Codec Driver
++=================
++
++The codec driver is generic and hardware independent code that configures the
++codec to provide audio capture and playback. It should contain no code that is
++specific to the target platform or machine. All platform and machine specific
++code should be added to the platform and machine drivers respectively.
++
++Each codec driver must provide the following features:-
++
++ 1) Digital audio interface (DAI) description
++ 2) Digital audio interface configuration
++ 3) PCM's description
++ 4) Codec control IO - using I2C, 3 Wire(SPI) or both API's
++ 5) Mixers and audio controls
++ 6) Sysclk configuration
++ 7) Codec audio operations
++
++Optionally, codec drivers can also provide:-
++
++ 8) DAPM description.
++ 9) DAPM event handler.
++10) DAC Digital mute control.
++
++It's probably best to use this guide in conjuction with the existing codec
++driver code in sound/soc/codecs/
++
++ASoC Codec driver breakdown
++===========================
++
++1 - Digital Audio Interface (DAI) description
++---------------------------------------------
++The DAI is a digital audio data transfer link between the codec and host SoC
++CPU. It typically has data transfer capabilities in both directions
++(playback and capture) and can run at a variety of different speeds.
++Supported interfaces currently include AC97, I2S and generic PCM style links.
++Please read DAI.txt for implementation information.
++
++
++2 - Digital Audio Interface (DAI) configuration
++-----------------------------------------------
++DAI configuration is handled by the codec_pcm_prepare function and is
++responsible for configuring and starting the DAI on the codec. This can be
++called multiple times and is atomic. It can access the runtime parameters.
++
++This usually consists of a large function with numerous switch statements to
++set up each configuration option. These options are set by the core at runtime.
++
++
++3 - Codec PCM's
++---------------
++Each codec must have it's PCM's defined. This defines the number of channels,
++stream names, callbacks and codec name. It is also used to register the DAI
++with the ASoC core. The PCM structure also associates the DAI capabilities with
++the ALSA PCM.
++
++e.g.
++
++static struct snd_soc_pcm_codec wm8731_pcm_client = {
++ .name = "WM8731",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = wm8731_config_sysclk,
++ .ops = {
++ .prepare = wm8731_pcm_prepare,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8731_hwfmt),
++ .modes = &wm8731_hwfmt[0],
++ },
++};
++
++
++4 - Codec control IO
++--------------------
++The codec can ususally be controlled via an I2C or SPI style interface (AC97
++combines control with data in the DAI). The codec drivers will have to provide
++functions to read and write the codec registers along with supplying a register
++cache:-
++
++ /* IO control data and register cache */
++ void *control_data; /* codec control (i2c/3wire) data */
++ void *reg_cache;
++
++Codec read/write should do any data formatting and call the hardware read write
++below to perform the IO. These functions are called by the core and alsa when
++performing DAPM or changing the mixer:-
++
++ unsigned int (*read)(struct snd_soc_codec *, unsigned int);
++ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
++
++Codec hardware IO functions - usually points to either the I2C, SPI or AC97
++read/write:-
++
++ hw_write_t hw_write;
++ hw_read_t hw_read;
++
++
++5 - Mixers and audio controls
++-----------------------------
++All the codec mixers and audio controls can be defined using the convenience
++macros defined in soc.h.
++
++ #define SOC_SINGLE(xname, reg, shift, mask, invert)
++
++Defines a single control as follows:-
++
++ xname = Control name e.g. "Playback Volume"
++ reg = codec register
++ shift = control bit(s) offset in register
++ mask = control bit size(s) e.g. mask of 7 = 3 bits
++ invert = the control is inverted
++
++Other macros include:-
++
++ #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
++
++A stereo control
++
++ #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
++
++A stereo control spanning 2 registers
++
++ #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
++
++Defines an single enumerated control as follows:-
++
++ xreg = register
++ xshift = control bit(s) offset in register
++ xmask = control bit(s) size
++ xtexts = pointer to array of strings that describe each setting
++
++ #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
++
++Defines a stereo enumerated control
++
++
++6 - System clock configuration.
++-------------------------------
++The system clock that drives the audio subsystem can change depending on sample
++rate and the system power state. i.e.
++
++o Higher sample rates sometimes need a higher system clock.
++o Low system power states can sometimes limit the available clocks.
++
++This function is a callback that the machine driver can call to set and
++determine if the clock and sample rate combination is supported by the codec at
++the present time (and system state).
++
++NOTE: If the codec has a PLL then it has a lot more flexability wrt clock and
++sample rate combinations.
++
++Your config_sysclock function should return the MCLK if it's a valid
++combination for your codec else 0;
++
++Please read clocking.txt now.
++
++
++7 - Codec Audio Operations
++--------------------------
++The codec driver also supports the following alsa operations:-
++
++/* SoC audio ops */
++struct snd_soc_ops {
++ int (*startup)(snd_pcm_substream_t *);
++ void (*shutdown)(snd_pcm_substream_t *);
++ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *);
++ int (*hw_free)(snd_pcm_substream_t *);
++ int (*prepare)(snd_pcm_substream_t *);
++};
++
++Please refer to the alsa driver PCM documentation for details.
++http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
++
++
++8 - DAPM description.
++---------------------
++The Dynamic Audio Power Management description describes the codec's power
++components, their relationships and registers to the ASoC core. Please read
++dapm.txt for details of building the description.
++
++Please also see the examples in other codec drivers.
++
++
++9 - DAPM event handler
++----------------------
++This function is a callback that handles codec domain PM calls and system
++domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep
++when not in use.
++
++Power states:-
++
++ SNDRV_CTL_POWER_D0: /* full On */
++ /* vref/mid, clk and osc on, active */
++
++ SNDRV_CTL_POWER_D1: /* partial On */
++ SNDRV_CTL_POWER_D2: /* partial On */
++
++ SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except vref/vmid, inactive */
++
++ SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
++
++
++10 - Codec DAC digital mute control.
++------------------------------------
++Most codecs have a digital mute before the DAC's that can be used to minimise
++any system noise. The mute stops any digital data from entering the DAC.
++
++A callback can be created that is called by the core for each codec DAI when the
++mute is applied or freed.
++
++i.e.
++
++static int wm8974_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
++ if(mute)
++ wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
++ else
++ wm8974_write(codec, WM8974_DAC, mute_reg);
++ return 0;
++}
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/dapm.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/dapm.txt
+@@ -0,0 +1,297 @@
++Dynamic Audio Power Management for Portable Devices
++===================================================
++
++1. Description
++==============
++
++Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices
++to use the minimum amount of power within the audio subsystem at all times. It
++is independent of other kernel PM and as such, can easily co-exist with the
++other PM systems.
++
++DAPM is also completely transparent to all user space applications as all power
++switching is done within the ASoC core. No code changes or recompiling are
++required for user space applications. DAPM makes power switching descisions based
++upon any audio stream (capture/playback) activity and audio mixer settings
++within the device.
++
++DAPM spans the whole machine. It covers power control within the entire audio
++subsystem, this includes internal codec power blocks and machine level power
++systems.
++
++There are 4 power domains within DAPM
++
++ 1. Codec domain - VREF, VMID (core codec and audio power)
++ Usually controlled at codec probe/remove and suspend/resume, although
++ can be set at stream time if power is not needed for sidetone, etc.
++
++ 2. Platform/Machine domain - physically connected inputs and outputs
++ Is platform/machine and user action specific, is configured by the
++ machine driver and responds to asynchronous events e.g when HP
++ are inserted
++
++ 3. Path domain - audio susbsystem signal paths
++ Automatically set when mixer and mux settings are changed by the user.
++ e.g. alsamixer, amixer.
++
++ 4. Stream domain - DAC's and ADC's.
++ Enabled and disabled when stream playback/capture is started and
++ stopped respectively. e.g. aplay, arecord.
++
++All DAPM power switching descisons are made automatically by consulting an audio
++routing map of the whole machine. This map is specific to each machine and
++consists of the interconnections between every audio component (including
++internal codec components). All audio components that effect power are called
++widgets hereafter.
++
++
++2. DAPM Widgets
++===============
++
++Audio DAPM widgets fall into a number of types:-
++
++ o Mixer - Mixes several analog signals into a single analog signal.
++ o Mux - An analog switch that outputs only 1 of it's inputs.
++ o PGA - A programmable gain amplifier or attenuation widget.
++ o ADC - Analog to Digital Converter
++ o DAC - Digital to Analog Converter
++ o Switch - An analog switch
++ o Input - A codec input pin
++ o Output - A codec output pin
++ o Headphone - Headphone (and optional Jack)
++ o Mic - Mic (and optional Jack)
++ o Line - Line Input/Output (and optional Jack)
++ o Speaker - Speaker
++ o Pre - Special PRE widget (exec before all others)
++ o Post - Special POST widget (exec after all others)
++
++(Widgets are defined in include/sound/soc-dapm.h)
++
++Widgets are usually added in the codec driver and the machine driver. There are
++convience macros defined in soc-dapm.h that can be used to quickly build a
++list of widgets of the codecs and machines DAPM widgets.
++
++Most widgets have a name, register, shift and invert. Some widgets have extra
++parameters for stream name and kcontrols.
++
++
++2.1 Stream Domain Widgets
++-------------------------
++
++Stream Widgets relate to the stream power domain and only consist of ADC's
++(analog to digital converters) and DAC's (digital to analog converters).
++
++Stream widgets have the following format:-
++
++SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
++
++NOTE: the stream name must match the corresponding stream name in your codecs
++snd_soc_codec_dai.
++
++e.g. stream widgets for HiFi playback and capture
++
++SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
++SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
++
++
++2.2 Path Domain Widgets
++-----------------------
++
++Path domain widgets have a ability to control or effect the audio signal or
++audio paths within the audio subsystem. They have the following form:-
++
++SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
++
++Any widget kcontrols can be set using the controls and num_controls members.
++
++e.g. Mixer widget (the kcontrols are declared first)
++
++/* Output Mixer */
++static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
++SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
++SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
++};
++
++SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
++ ARRAY_SIZE(wm8731_output_mixer_controls)),
++
++
++2.3 Platform/Machine domain Widgets
++-----------------------------------
++
++Machine widgets are different from codec widgets in that they don't have a
++codec register bit associated with them. A machine widget is assigned to each
++machine audio component (non codec) that can be independently powered. e.g.
++
++ o Speaker Amp
++ o Microphone Bias
++ o Jack connectors
++
++A machine widget can have an optional call back.
++
++e.g. Jack connector widget for an external Mic that enables Mic Bias
++when the Mic is inserted:-
++
++static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
++{
++ if(SND_SOC_DAPM_EVENT_ON(event))
++ set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
++ else
++ reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
++
++ return 0;
++}
++
++SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
++
++
++2.4 Codec Domain
++----------------
++
++The Codec power domain has no widgets and is handled by the codecs DAPM event
++handler. This handler is called when the codec powerstate is changed wrt to any
++stream event or by kernel PM events.
++
++
++2.5 Virtual Widgets
++-------------------
++
++Sometimes widgets exist in the codec or machine audio map that don't have any
++corresponding register bit for power control. In this case it's necessary to
++create a virtual widget - a widget with no control bits e.g.
++
++SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
++
++This can be used to merge to signal paths together in software.
++
++After all the widgets have been defined, they can then be added to the DAPM
++subsystem individually with a call to snd_soc_dapm_new_control().
++
++
++3. Codec Widget Interconnections
++================================
++
++Widgets are connected to each other within the codec and machine by audio
++paths (called interconnections). Each interconnection must be defined in order
++to create a map of all audio paths between widgets.
++This is easiest with a diagram of the codec (and schematic of the machine audio
++system), as it requires joining widgets together via their audio signal paths.
++
++i.e. from the WM8731 codec's output mixer (wm8731.c)
++
++The WM8731 output mixer has 3 inputs (sources)
++
++ 1. Line Bypass Input
++ 2. DAC (HiFi playback)
++ 3. Mic Sidetone Input
++
++Each input in this example has a kcontrol associated with it (defined in example
++above) and is connected to the output mixer via it's kcontrol name. We can now
++connect the destination widget (wrt audio signal) with it's source widgets.
++
++ /* output mixer */
++ {"Output Mixer", "Line Bypass Switch", "Line Input"},
++ {"Output Mixer", "HiFi Playback Switch", "DAC"},
++ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
++
++So we have :-
++
++ Destination Widget <=== Path Name <=== Source Widget
++
++Or:-
++
++ Sink, Path, Source
++
++Or :-
++
++ "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".
++
++When there is no path name connecting widgets (e.g. a direct connection) we
++pass NULL for the path name.
++
++Interconnections are created with a call to:-
++
++snd_soc_dapm_connect_input(codec, sink, path, source);
++
++Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
++interconnections have been registered with the core. This causes the core to
++scan the codec and machine so that the internal DAPM state matches the
++physical state of the machine.
++
++
++3.1 Machine Widget Interconnections
++-----------------------------------
++Machine widget interconnections are created in the same way as codec ones and
++directly connect the codec pins to machine level widgets.
++
++e.g. connects the speaker out codec pins to the internal speaker.
++
++ /* ext speaker connected to codec pins LOUT2, ROUT2 */
++ {"Ext Spk", NULL , "ROUT2"},
++ {"Ext Spk", NULL , "LOUT2"},
++
++This allows the DAPM to power on and off pins that are connected (and in use)
++and pins that are NC respectively.
++
++
++4 Endpoint Widgets
++===================
++An endpoint is a start or end point (widget) of an audio signal within the
++machine and includes the codec. e.g.
++
++ o Headphone Jack
++ o Internal Speaker
++ o Internal Mic
++ o Mic Jack
++ o Codec Pins
++
++When a codec pin is NC it can be marked as not used with a call to
++
++snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
++
++The last argument is 0 for inactive and 1 for active. This way the pin and its
++input widget will never be powered up and consume power.
++
++This also applies to machine widgets. e.g. if a headphone is connected to a
++jack then the jack can be marked active. If the headphone is removed, then
++the headphone jack can be marked inactive.
++
++
++5 DAPM Widget Events
++====================
++
++Some widgets can register their interest with the DAPM core in PM events.
++e.g. A Speaker with an amplifier registers a widget so the amplifier can be
++powered only when the spk is in use.
++
++/* turn speaker amplifier on/off depending on use */
++static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
++{
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
++ else
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
++
++ return 0;
++}
++
++/* corgi machine dapm widgets */
++static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
++ SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
++
++Please see soc-dapm.h for all other widgets that support events.
++
++
++5.1 Event types
++---------------
++
++The following event types are supported by event widgets.
++
++/* dapm event types */
++#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
++#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
++#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
++#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
++#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
++#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/machine.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/machine.txt
+@@ -0,0 +1,114 @@
++ASoC Machine Driver
++===================
++
++The ASoC machine (or board) driver is the code that glues together the platform
++and codec drivers.
++
++The machine driver can contain codec and platform specific code. It registers
++the audio subsystem with the kernel as a platform device and is represented by
++the following struct:-
++
++/* SoC machine */
++struct snd_soc_machine {
++ char *name;
++
++ int (*probe)(struct platform_device *pdev);
++ int (*remove)(struct platform_device *pdev);
++
++ /* the pre and post PM functions are used to do any PM work before and
++ * after the codec and DAI's do any PM work. */
++ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
++ int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
++ int (*resume_pre)(struct platform_device *pdev);
++ int (*resume_post)(struct platform_device *pdev);
++
++ /* machine stream operations */
++ struct snd_soc_ops *ops;
++
++ /* CPU <--> Codec DAI links */
++ struct snd_soc_dai_link *dai_link;
++ int num_links;
++};
++
++probe()/remove()
++----------------
++probe/remove are optional. Do any machine specific probe here.
++
++
++suspend()/resume()
++------------------
++The machine driver has pre and post versions of suspend and resume to take care
++of any machine audio tasks that have to be done before or after the codec, DAI's
++and DMA is suspended and resumed. Optional.
++
++
++Machine operations
++------------------
++The machine specific audio operations can be set here. Again this is optional.
++
++
++Machine DAI Configuration
++-------------------------
++The machine DAI configuration glues all the codec and CPU DAI's together. It can
++also be used to set up the DAI system clock and for any machine related DAI
++initialisation e.g. the machine audio map can be connected to the codec audio
++map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c
++for examples.
++
++struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
++
++/* corgi digital audio interface glue - connects codec <--> CPU */
++static struct snd_soc_dai_link corgi_dai = {
++ .name = "WM8731",
++ .stream_name = "WM8731",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8731_dai,
++ .init = corgi_wm8731_init,
++ .config_sysclk = corgi_config_sysclk,
++};
++
++struct snd_soc_machine then sets up the machine with it's DAI's. e.g.
++
++/* corgi audio machine driver */
++static struct snd_soc_machine snd_soc_machine_corgi = {
++ .name = "Corgi",
++ .dai_link = &corgi_dai,
++ .num_links = 1,
++ .ops = &corgi_ops,
++};
++
++
++Machine Audio Subsystem
++-----------------------
++
++The machine soc device glues the platform, machine and codec driver together.
++Private data can also be set here. e.g.
++
++/* corgi audio private data */
++static struct wm8731_setup_data corgi_wm8731_setup = {
++ .i2c_address = 0x1b,
++};
++
++/* corgi audio subsystem */
++static struct snd_soc_device corgi_snd_devdata = {
++ .machine = &snd_soc_machine_corgi,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8731,
++ .codec_data = &corgi_wm8731_setup,
++};
++
++
++Machine Power Map
++-----------------
++
++The machine driver can optionally extend the codec power map and to become an
++audio power map of the audio subsystem. This allows for automatic power up/down
++of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
++sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
++details.
++
++
++Machine Controls
++----------------
++
++Machine specific audio mixer controls can be added in the dai init function.
+\ No newline at end of file
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/overview.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/overview.txt
+@@ -0,0 +1,83 @@
++ALSA SoC Layer
++==============
++
++The overall project goal of the ALSA System on Chip (ASoC) layer is to provide
++better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00,
++iMX, etc) and portable audio codecs. Currently there is some support in the
++kernel for SoC audio, however it has some limitations:-
++
++ * Currently, codec drivers are often tightly coupled to the underlying SoC
++ cpu. This is not ideal and leads to code duplication i.e. Linux now has 4
++ different wm8731 drivers for 4 different SoC platforms.
++
++ * There is no standard method to signal user initiated audio events.
++ e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion
++ event. These are quite common events on portable devices and ofter require
++ machine specific code to re route audio, enable amps etc after such an event.
++
++ * Current drivers tend to power up the entire codec when playing
++ (or recording) audio. This is fine for a PC, but tends to waste a lot of
++ power on portable devices. There is also no support for saving power via
++ changing codec oversampling rates, bias currents, etc.
++
++
++ASoC Design
++===========
++
++The ASoC layer is designed to address these issues and provide the following
++features :-
++
++ * Codec independence. Allows reuse of codec drivers on other platforms
++ and machines.
++
++ * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface
++ and codec registers it's audio interface capabilities with the core and are
++ subsequently matched and configured when the application hw params are known.
++
++ * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
++ it's minimum power state at all times. This includes powering up/down
++ internal power blocks depending on the internal codec audio routing and any
++ active streams.
++
++ * Pop and click reduction. Pops and clicks can be reduced by powering the
++ codec up/down in the correct sequence (including using digital mute). ASoC
++ signals the codec when to change power states.
++
++ * Machine specific controls: Allow machines to add controls to the sound card
++ e.g. volume control for speaker amp.
++
++To achieve all this, ASoC basically splits an embedded audio system into 3
++components :-
++
++ * Codec driver: The codec driver is platform independent and contains audio
++ controls, audio interface capabilities, codec dapm definition and codec IO
++ functions.
++
++ * Platform driver: The platform driver contains the audio dma engine and audio
++ interface drivers (e.g. I2S, AC97, PCM) for that platform.
++
++ * Machine driver: The machine driver handles any machine specific controls and
++ audio events. i.e. turing on an amp at start of playback.
++
++
++Documentation
++=============
++
++The documentation is spilt into the following sections:-
++
++overview.txt: This file.
++
++codec.txt: Codec driver internals.
++
++DAI.txt: Description of Digital Audio Interface standards and how to configure
++a DAI within your codec and CPU DAI drivers.
++
++dapm.txt: Dynamic Audio Power Management
++
++platform.txt: Platform audio DMA and DAI.
++
++machine.txt: Machine driver internals.
++
++pop_clicks.txt: How to minimise audio artifacts.
++
++clocking.txt: ASoC clocking for best power performance.
+\ No newline at end of file
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/platform.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/platform.txt
+@@ -0,0 +1,58 @@
++ASoC Platform Driver
++====================
++
++An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
++and control. The platform drivers only target the SoC CPU and must have no board
++specific code.
++
++Audio DMA
++=========
++
++The platform DMA driver optionally supports the following alsa operations:-
++
++/* SoC audio ops */
++struct snd_soc_ops {
++ int (*startup)(snd_pcm_substream_t *);
++ void (*shutdown)(snd_pcm_substream_t *);
++ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *);
++ int (*hw_free)(snd_pcm_substream_t *);
++ int (*prepare)(snd_pcm_substream_t *);
++ int (*trigger)(snd_pcm_substream_t *, int);
++};
++
++The platform driver exports it's DMA functionailty via struct snd_soc_platform:-
++
++struct snd_soc_platform {
++ char *name;
++
++ int (*probe)(struct platform_device *pdev);
++ int (*remove)(struct platform_device *pdev);
++ int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
++ int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
++
++ /* pcm creation and destruction */
++ int (*pcm_new)(snd_card_t *, struct snd_soc_codec_dai *, snd_pcm_t *);
++ void (*pcm_free)(snd_pcm_t *);
++
++ /* platform stream ops */
++ snd_pcm_ops_t *pcm_ops;
++};
++
++Please refer to the alsa driver documentation for details of audio DMA.
++http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
++
++An example DMA driver is soc/pxa/pxa2xx-pcm.c
++
++
++SoC DAI Drivers
++===============
++
++Each SoC DAI driver must provide the following features:-
++
++ 1) Digital audio interface (DAI) description
++ 2) Digital audio interface configuration
++ 3) PCM's description
++ 4) Sysclk configuration
++ 5) Suspend and resume (optional)
++
++Please see codec.txt for a description of items 1 - 4.
+Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/pops_clicks.txt
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/pops_clicks.txt
+@@ -0,0 +1,52 @@
++Audio Pops and Clicks
++=====================
++
++Pops and clicks are unwanted audio artifacts caused by the powering up and down
++of components within the audio subsystem. This is noticable on PC's when an audio
++module is either loaded or unloaded (at module load time the sound card is
++powered up and causes a popping noise on the speakers).
++
++Pops and clicks can be more frequent on portable systems with DAPM. This is because
++the components within the subsystem are being dynamically powered depending on
++the audio usage and this can subsequently cause a small pop or click every time a
++component power state is changed.
++
++
++Minimising Playback Pops and Clicks
++===================================
++
++Playback pops in portable audio subsystems cannot be completely eliminated atm,
++however future audio codec hardware will have better pop and click supression.
++Pops can be reduced within playback by powering the audio components in a
++specific order. This order is different for startup and shutdown and follows
++some basic rules:-
++
++ Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
++
++ Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
++
++This assumes that the codec PCM output path from the DAC is via a mixer and then
++a PGA (programmable gain amplifier) before being output to the speakers.
++
++
++Minimising Capture Pops and Clicks
++==================================
++
++Capture artifacts are somewhat easier to get rid as we can delay activating the
++ADC until all the pops have occured. This follows similar power rules to
++playback in that components are powered in a sequence depending upon stream
++startup or shutdown.
++
++ Startup Order - Input PGA --> Mixers --> ADC
++
++ Shutdown Order - ADC --> Mixers --> Input PGA
++
++
++Zipper Noise
++============
++An unwanted zipper noise can occur within the audio playback or capture stream
++when a volume control is changed near its maximum gain value. The zipper noise
++is heard when the gain increase or decrease changes the mean audio signal
++amplitude too quickly. It can be minimised by enabling the zero cross setting
++for each volume control. The ZC forces the gain change to occur when the signal
++crosses the zero amplitude line.
+Index: linux-2.6-pxa-new/include/sound/ac97_codec.h
+===================================================================
+--- linux-2.6-pxa-new.orig/include/sound/ac97_codec.h
++++ linux-2.6-pxa-new/include/sound/ac97_codec.h
+@@ -425,6 +425,7 @@ struct snd_ac97_build_ops {
+
+ struct snd_ac97_bus_ops {
+ void (*reset) (struct snd_ac97 *ac97);
++ void (*warm_reset)(struct snd_ac97 *ac97);
+ void (*write) (struct snd_ac97 *ac97, unsigned short reg, unsigned short val);
+ unsigned short (*read) (struct snd_ac97 *ac97, unsigned short reg);
+ void (*wait) (struct snd_ac97 *ac97);
+Index: linux-2.6-pxa-new/include/sound/soc-dapm.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/include/sound/soc-dapm.h
+@@ -0,0 +1,286 @@
++/*
++ * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management
++ *
++ * Author: Liam Girdwood
++ * Created: Aug 11th 2005
++ * Copyright: Wolfson Microelectronics. PLC.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef __LINUX_SND_SOC_DAPM_H
++#define __LINUX_SND_SOC_DAPM_H
++
++#include <linux/device.h>
++#include <linux/types.h>
++#include <sound/control.h>
++#include <sound/soc.h>
++
++/* widget has no PM register bit */
++#define SND_SOC_NOPM -1
++
++/*
++ * SoC dynamic audio power managment
++ *
++ * We can have upto 4 power domains
++ * 1. Codec domain - VREF, VMID
++ * Usually controlled at codec probe/remove, although can be set
++ * at stream time if power is not needed for sidetone, etc.
++ * 2. Platform/Machine domain - physically connected inputs and outputs
++ * Is platform/machine and user action specific, is set in the machine
++ * driver and by userspace e.g when HP are inserted
++ * 3. Path domain - Internal codec path mixers
++ * Are automatically set when mixer and mux settings are
++ * changed by the user.
++ * 4. Stream domain - DAC's and ADC's.
++ * Enabled when stream playback/capture is started.
++ */
++
++/* codec domain */
++#define SND_SOC_DAPM_VMID(wname) \
++{ .id = snd_soc_dapm_vmid, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0}
++
++/* platform domain */
++#define SND_SOC_DAPM_INPUT(wname) \
++{ .id = snd_soc_dapm_input, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0}
++#define SND_SOC_DAPM_OUTPUT(wname) \
++{ .id = snd_soc_dapm_output, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0}
++#define SND_SOC_DAPM_MIC(wname, wevent) \
++{ .id = snd_soc_dapm_mic, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0, .event = wevent, \
++ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
++#define SND_SOC_DAPM_HP(wname, wevent) \
++{ .id = snd_soc_dapm_hp, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0, .event = wevent, \
++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
++#define SND_SOC_DAPM_SPK(wname, wevent) \
++{ .id = snd_soc_dapm_spk, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0, .event = wevent, \
++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
++#define SND_SOC_DAPM_LINE(wname, wevent) \
++{ .id = snd_soc_dapm_line, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0, .event = wevent, \
++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
++
++/* path domain */
++#define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\
++ wcontrols, wncontrols) \
++{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols}
++#define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \
++ wcontrols, wncontrols)\
++{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols}
++#define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
++{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0}
++#define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \
++{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1}
++#define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \
++{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1}
++
++/* path domain with event - event handler must return 0 for success */
++#define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
++ wncontrols, wevent, wflags) \
++{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \
++ .event = wevent, .event_flags = wflags}
++#define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \
++ wncontrols, wevent, wflags) \
++{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \
++ .event = wevent, .event_flags = wflags}
++#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \
++{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0, \
++ .event = wevent, .event_flags = wflags}
++#define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \
++ wevent, wflags) \
++{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1 \
++ .event = wevent, .event_flags = wflags}
++#define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
++ wevent, wflags) \
++{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \
++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \
++ .event = wevent, .event_flags = wflags}
++
++/* events that are pre and post DAPM */
++#define SND_SOC_DAPM_PRE(wname, wevent) \
++{ .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0, .event = wevent, \
++ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD}
++#define SND_SOC_DAPM_POST(wname, wevent) \
++{ .id = snd_soc_dapm_post, .name = wname, .kcontrols = NULL, \
++ .num_kcontrols = 0, .event = wevent, \
++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
++
++/* stream domain */
++#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
++{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
++ .shift = wshift, .invert = winvert}
++#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
++{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
++ .shift = wshift, .invert = winvert}
++
++/* dapm kcontrol types */
++#define SOC_DAPM_SINGLE(xname, reg, shift, mask, invert) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++ .info = snd_soc_info_volsw, \
++ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
++ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) }
++#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, mask, invert, \
++ power) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
++ .info = snd_soc_info_volsw, \
++ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
++ .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\
++ ((mask) << 16) | ((invert) << 24) }
++#define SOC_DAPM_ENUM(xname, xenum) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++ .info = snd_soc_info_enum_double, \
++ .get = snd_soc_dapm_get_enum_double, \
++ .put = snd_soc_dapm_put_enum_double, \
++ .private_value = (unsigned long)&xenum }
++
++/* dapm stream operations */
++#define SND_SOC_DAPM_STREAM_NOP 0x0
++#define SND_SOC_DAPM_STREAM_START 0x1
++#define SND_SOC_DAPM_STREAM_STOP 0x2
++#define SND_SOC_DAPM_STREAM_SUSPEND 0x4
++#define SND_SOC_DAPM_STREAM_RESUME 0x8
++#define SND_SOC_DAPM_STREAM_PAUSE_PUSH 0x10
++#define SND_SOC_DAPM_STREAM_PAUSE_RELEASE 0x20
++
++/* dapm event types */
++#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
++#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
++#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
++#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
++#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
++#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
++
++/* convenience event type detection */
++#define SND_SOC_DAPM_EVENT_ON(e) \
++ (e & (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU))
++#define SND_SOC_DAPM_EVENT_OFF(e) \
++ (e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD))
++
++struct snd_soc_dapm_widget;
++enum snd_soc_dapm_type;
++struct snd_soc_dapm_path;
++struct snd_soc_dapm_pin;
++
++/* dapm controls */
++int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
++ const struct snd_soc_dapm_widget *widget);
++
++/* dapm path setup */
++int snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
++ const char *sink_name, const char *control_name, const char *src_name);
++int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
++void snd_soc_dapm_free(struct snd_soc_device *socdev);
++
++/* dapm events */
++int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
++ int event);
++
++/* dapm sys fs - used by the core */
++int snd_soc_dapm_sys_add(struct device *dev);
++
++/* dapm audio endpoint control */
++int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
++ char *pin, int status);
++int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
++
++/* dapm widget types */
++enum snd_soc_dapm_type {
++ snd_soc_dapm_input = 0, /* input pin */
++ snd_soc_dapm_output, /* output pin */
++ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */
++ snd_soc_dapm_mixer, /* mixes several analog signals together */
++ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */
++ snd_soc_dapm_adc, /* analog to digital converter */
++ snd_soc_dapm_dac, /* digital to analog converter */
++ snd_soc_dapm_micbias, /* microphone bias (power) */
++ snd_soc_dapm_mic, /* microphone */
++ snd_soc_dapm_hp, /* headphones */
++ snd_soc_dapm_spk, /* speaker */
++ snd_soc_dapm_line, /* line input/output */
++ snd_soc_dapm_switch, /* analog switch */
++ snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */
++ snd_soc_dapm_pre, /* machine specific pre widget - exec first */
++ snd_soc_dapm_post, /* machine specific post widget - exec last */
++};
++
++/* dapm audio path between two widgets */
++struct snd_soc_dapm_path {
++ char *name;
++ char *long_name;
++
++ /* source (input) and sink (output) widgets */
++ struct snd_soc_dapm_widget *source;
++ struct snd_soc_dapm_widget *sink;
++ struct snd_kcontrol *kcontrol;
++
++ /* status */
++ u32 connect:1; /* source and sink widgets are connected */
++ u32 walked:1; /* path has been walked */
++
++ struct list_head list_source;
++ struct list_head list_sink;
++ struct list_head list;
++};
++
++/* dapm widget */
++struct snd_soc_dapm_widget {
++ enum snd_soc_dapm_type id;
++ char *name; /* widget name */
++ char *sname; /* stream name */
++ struct snd_soc_codec *codec;
++ struct list_head list;
++
++ /* dapm control */
++ short reg; /* negative reg = no direct dapm */
++ unsigned char shift; /* bits to shift */
++ unsigned int saved_value; /* widget saved value */
++ unsigned int value; /* widget current value */
++ unsigned char power:1; /* block power status */
++ unsigned char invert:1; /* invert the power bit */
++ unsigned char active:1; /* active stream on DAC, ADC's */
++ unsigned char connected:1; /* connected codec pin */
++ unsigned char new:1; /* cnew complete */
++ unsigned char ext:1; /* has external widgets */
++ unsigned char muted:1; /* muted for pop reduction */
++ unsigned char suspend:1; /* was active before suspend */
++ unsigned char pmdown:1; /* waiting for timeout */
++
++ /* external events */
++ unsigned short event_flags; /* flags to specify event types */
++ int (*event)(struct snd_soc_dapm_widget*, int);
++
++ /* kcontrols that relate to this widget */
++ int num_kcontrols;
++ const struct snd_kcontrol_new *kcontrols;
++
++ /* widget input and outputs */
++ struct list_head sources;
++ struct list_head sinks;
++};
++
++#endif
+Index: linux-2.6-pxa-new/include/sound/soc.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/include/sound/soc.h
+@@ -0,0 +1,487 @@
++/*
++ * linux/sound/soc.h -- ALSA SoC Layer
++ *
++ * Author: Liam Girdwood
++ * Created: Aug 11th 2005
++ * Copyright: Wolfson Microelectronics. PLC.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef __LINUX_SND_SOC_H
++#define __LINUX_SND_SOC_H
++
++#include <linux/platform_device.h>
++#include <linux/types.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/control.h>
++#include <sound/ac97_codec.h>
++
++#define SND_SOC_VERSION "0.12.4"
++
++/*
++ * Convenience kcontrol builders
++ */
++#define SOC_SINGLE_VALUE(reg,shift,mask,invert) ((reg) | ((shift) << 8) |\
++ ((shift) << 12) | ((mask) << 16) | ((invert) << 24))
++#define SOC_SINGLE_VALUE_EXT(reg,mask,invert) ((reg) | ((mask) << 16) |\
++ ((invert) << 31))
++#define SOC_SINGLE(xname, reg, shift, mask, invert) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
++ .put = snd_soc_put_volsw, \
++ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) }
++#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
++ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
++ .put = snd_soc_put_volsw, \
++ .private_value = (reg) | ((shift_left) << 8) | \
++ ((shift_right) << 12) | ((mask) << 16) | ((invert) << 24) }
++#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
++ .info = snd_soc_info_volsw_2r, \
++ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
++ .private_value = (reg_left) | ((shift) << 8) | \
++ ((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) }
++#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \
++{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
++ .mask = xmask, .texts = xtexts }
++#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \
++ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts)
++#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \
++{ .mask = xmask, .texts = xtexts }
++#define SOC_ENUM(xname, xenum) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\
++ .info = snd_soc_info_enum_double, \
++ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \
++ .private_value = (unsigned long)&xenum }
++#define SOC_SINGLE_EXT(xname, xreg, xmask, xinvert,\
++ xhandler_get, xhandler_put) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++ .info = snd_soc_info_volsw_ext, \
++ .get = xhandler_get, .put = xhandler_put, \
++ .private_value = SOC_SINGLE_VALUE_EXT(xreg, xmask, xinvert) }
++#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++ .info = snd_soc_info_bool_ext, \
++ .get = xhandler_get, .put = xhandler_put, \
++ .private_value = xdata }
++#define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++ .info = snd_soc_info_enum_ext, \
++ .get = xhandler_get, .put = xhandler_put, \
++ .private_value = (unsigned long)&xenum }
++
++/*
++ * Digital Audio Interface (DAI) types
++ */
++#define SND_SOC_DAI_AC97 0x1
++#define SND_SOC_DAI_I2S 0x2
++#define SND_SOC_DAI_PCM 0x4
++
++/*
++ * DAI hardware audio formats
++ */
++#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */
++#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */
++#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */
++#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM or LRC */
++#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM or LRC */
++#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */
++
++/*
++ * DAI hardware signal inversions
++ */
++#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */
++#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */
++#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */
++#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */
++
++/*
++ * DAI hardware clock masters
++ * This is wrt the codec, the inverse is true for the interface
++ * i.e. if the codec is clk and frm master then the interface is
++ * clk and frame slave.
++ */
++#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */
++#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */
++#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */
++#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */
++
++#define SND_SOC_DAIFMT_FORMAT_MASK 0x00ff
++#define SND_SOC_DAIFMT_INV_MASK 0x0f00
++#define SND_SOC_DAIFMT_CLOCK_MASK 0xf000
++
++/*
++ * DAI hardware audio direction
++ */
++#define SND_SOC_DAIDIR_PLAYBACK 0x1
++#define SND_SOC_DAIDIR_CAPTURE 0x2
++
++/*
++ * DAI hardware Time Division Multiplexing (TDM) Slots
++ * Left and Right data word positions
++ * This is measured in words (sample size) and not bits.
++ */
++#define SND_SOC_DAITDM_LRDW(l,r) ((l << 8) | r)
++
++/*
++ * DAI hardware clock ratios
++ * bit clock can either be a generated by dividing mclk or
++ * by multiplying sample rate, hence there are 2 definitions below
++ * depending on codec type.
++ */
++/* ratio of sample rate to mclk/sysclk */
++#define SND_SOC_FS_ALL 0xffff /* all mclk supported */
++
++/* bit clock dividers */
++#define SND_SOC_FSBD(x) (1 << (x - 1)) /* ratio mclk:bclk */
++#define SND_SOC_FSBD_REAL(x) (ffs(x))
++
++/* bit clock ratio to (sample rate * channels * word size) */
++#define SND_SOC_FSBW(x) (1 << (x - 1))
++#define SND_SOC_FSBW_REAL(x) (ffs(x))
++/* all bclk ratios supported */
++#define SND_SOC_FSB_ALL ~0ULL
++
++/*
++ * DAI hardware flags
++ */
++/* use bfs mclk divider mode (BCLK = MCLK / x) */
++#define SND_SOC_DAI_BFS_DIV 0x1
++/* use bfs rate mulitplier (BCLK = RATE * x)*/
++#define SND_SOC_DAI_BFS_RATE 0x2
++/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */
++#define SND_SOC_DAI_BFS_RCW 0x4
++/* capture and playback can use different clocks */
++#define SND_SOC_DAI_ASYNC 0x8
++/* can use gated BCLK */
++#define SND_SOC_DAI_GATED 0x10
++
++/*
++ * AC97 codec ID's bitmask
++ */
++#define SND_SOC_DAI_AC97_ID0 (1 << 0)
++#define SND_SOC_DAI_AC97_ID1 (1 << 1)
++#define SND_SOC_DAI_AC97_ID2 (1 << 2)
++#define SND_SOC_DAI_AC97_ID3 (1 << 3)
++
++struct snd_soc_device;
++struct snd_soc_pcm_stream;
++struct snd_soc_ops;
++struct snd_soc_dai_mode;
++struct snd_soc_pcm_runtime;
++struct snd_soc_codec_dai;
++struct snd_soc_cpu_dai;
++struct snd_soc_codec;
++struct snd_soc_machine_config;
++struct soc_enum;
++struct snd_soc_ac97_ops;
++struct snd_soc_clock_info;
++
++typedef int (*hw_write_t)(void *,const char* ,int);
++typedef int (*hw_read_t)(void *,char* ,int);
++
++extern struct snd_ac97_bus_ops soc_ac97_ops;
++
++/* pcm <-> DAI connect */
++void snd_soc_free_pcms(struct snd_soc_device *socdev);
++int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
++int snd_soc_register_card(struct snd_soc_device *socdev);
++
++/* set runtime hw params */
++int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
++ const struct snd_pcm_hardware *hw);
++int snd_soc_get_rate(int rate);
++
++/* codec IO */
++#define snd_soc_read(codec, reg) codec->read(codec, reg)
++#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
++
++/* codec register bit access */
++int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
++ unsigned short mask, unsigned short value);
++int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
++ unsigned short mask, unsigned short value);
++
++int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
++ struct snd_ac97_bus_ops *ops, int num);
++void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
++
++/*
++ *Controls
++ */
++struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
++ void *data, char *long_name);
++int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_info *uinfo);
++int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_info *uinfo);
++int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_info *uinfo);
++int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_info *uinfo);
++int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_info *uinfo);
++int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_info *uinfo);
++int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol);
++
++/* SoC PCM stream information */
++struct snd_soc_pcm_stream {
++ char *stream_name;
++ unsigned int rate_min; /* min rate */
++ unsigned int rate_max; /* max rate */
++ unsigned int channels_min; /* min channels */
++ unsigned int channels_max; /* max channels */
++ unsigned int active:1; /* stream is in use */
++};
++
++/* SoC audio ops */
++struct snd_soc_ops {
++ int (*startup)(struct snd_pcm_substream *);
++ void (*shutdown)(struct snd_pcm_substream *);
++ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
++ int (*hw_free)(struct snd_pcm_substream *);
++ int (*prepare)(struct snd_pcm_substream *);
++ int (*trigger)(struct snd_pcm_substream *, int);
++};
++
++/* SoC DAI hardware mode */
++struct snd_soc_dai_mode {
++ u16 fmt; /* SND_SOC_DAIFMT_* */
++ u16 tdm; /* SND_SOC_HWTDM_* */
++ u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */
++ u16 pcmrate; /* SND_SOC_HWRATE_* */
++ u16 pcmdir:2; /* SND_SOC_HWDIR_* */
++ u16 flags:8; /* hw flags */
++ u16 fs; /* mclk to rate divider */
++ u64 bfs; /* mclk to bclk dividers */
++ unsigned long priv; /* private mode data */
++};
++
++/* DAI capabilities */
++struct snd_soc_dai_cap {
++ int num_modes; /* number of DAI modes */
++ struct snd_soc_dai_mode *mode; /* array of supported DAI modes */
++};
++
++/* SoC Codec DAI */
++struct snd_soc_codec_dai {
++ char *name;
++ int id;
++
++ /* DAI capabilities */
++ struct snd_soc_pcm_stream playback;
++ struct snd_soc_pcm_stream capture;
++ struct snd_soc_dai_cap caps;
++
++ /* DAI runtime info */
++ struct snd_soc_dai_mode dai_runtime;
++ struct snd_soc_ops ops;
++ unsigned int (*config_sysclk)(struct snd_soc_codec_dai*,
++ struct snd_soc_clock_info *info, unsigned int clk);
++ int (*digital_mute)(struct snd_soc_codec *,
++ struct snd_soc_codec_dai*, int);
++ unsigned int mclk; /* the audio master clock */
++ unsigned int pll_in; /* the PLL input clock */
++ unsigned int pll_out; /* the PLL output clock */
++ unsigned int clk_div; /* internal clock divider << 1 (for fractions) */
++ unsigned int active;
++ unsigned char pop_wait:1;
++
++ /* DAI private data */
++ void *private_data;
++};
++
++/* SoC CPU DAI */
++struct snd_soc_cpu_dai {
++
++ /* DAI description */
++ char *name;
++ unsigned int id;
++ unsigned char type;
++
++ /* DAI callbacks */
++ int (*probe)(struct platform_device *pdev);
++ void (*remove)(struct platform_device *pdev);
++ int (*suspend)(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *cpu_dai);
++ int (*resume)(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *cpu_dai);
++ unsigned int (*config_sysclk)(struct snd_soc_cpu_dai *cpu_dai,
++ struct snd_soc_clock_info *info, unsigned int clk);
++
++ /* DAI capabilities */
++ struct snd_soc_pcm_stream capture;
++ struct snd_soc_pcm_stream playback;
++ struct snd_soc_dai_cap caps;
++
++ /* DAI runtime info */
++ struct snd_soc_dai_mode dai_runtime;
++ struct snd_soc_ops ops;
++ struct snd_pcm_runtime *runtime;
++ unsigned char active:1;
++ unsigned int mclk;
++ void *dma_data;
++
++ /* DAI private data */
++ void *private_data;
++};
++
++/* SoC Audio Codec */
++struct snd_soc_codec {
++ char *name;
++ struct module *owner;
++ struct mutex mutex;
++
++ /* callbacks */
++ int (*dapm_event)(struct snd_soc_codec *codec, int event);
++
++ /* runtime */
++ struct snd_card *card;
++ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
++ unsigned int active;
++ unsigned int pcm_devs;
++ void *private_data;
++
++ /* codec IO */
++ void *control_data; /* codec control (i2c/3wire) data */
++ unsigned int (*read)(struct snd_soc_codec *, unsigned int);
++ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
++ hw_write_t hw_write;
++ hw_read_t hw_read;
++ void *reg_cache;
++ short reg_cache_size;
++ short reg_cache_step;
++
++ /* dapm */
++ struct list_head dapm_widgets;
++ struct list_head dapm_paths;
++ unsigned int dapm_state;
++ unsigned int suspend_dapm_state;
++
++ /* codec DAI's */
++ struct snd_soc_codec_dai *dai;
++ unsigned int num_dai;
++};
++
++/* codec device */
++struct snd_soc_codec_device {
++ int (*probe)(struct platform_device *pdev);
++ int (*remove)(struct platform_device *pdev);
++ int (*suspend)(struct platform_device *pdev, pm_message_t state);
++ int (*resume)(struct platform_device *pdev);
++};
++
++/* SoC platform interface */
++struct snd_soc_platform {
++ char *name;
++
++ int (*probe)(struct platform_device *pdev);
++ int (*remove)(struct platform_device *pdev);
++ int (*suspend)(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *cpu_dai);
++ int (*resume)(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *cpu_dai);
++
++ /* pcm creation and destruction */
++ int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *,
++ struct snd_pcm *);
++ void (*pcm_free)(struct snd_pcm *);
++
++ /* platform stream ops */
++ struct snd_pcm_ops *pcm_ops;
++};
++
++/* SoC machine DAI configuration, glues a codec and cpu DAI together */
++struct snd_soc_dai_link {
++ char *name; /* Codec name */
++ char *stream_name; /* Stream name */
++
++ /* DAI */
++ struct snd_soc_codec_dai *codec_dai;
++ struct snd_soc_cpu_dai *cpu_dai;
++ u32 flags; /* DAI config preference flags */
++
++ /* codec/machine specific init - e.g. add machine controls */
++ int (*init)(struct snd_soc_codec *codec);
++
++ /* audio sysclock configuration */
++ unsigned int (*config_sysclk)(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info);
++};
++
++/* SoC machine */
++struct snd_soc_machine {
++ char *name;
++
++ int (*probe)(struct platform_device *pdev);
++ int (*remove)(struct platform_device *pdev);
++
++ /* the pre and post PM functions are used to do any PM work before and
++ * after the codec and DAI's do any PM work. */
++ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
++ int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
++ int (*resume_pre)(struct platform_device *pdev);
++ int (*resume_post)(struct platform_device *pdev);
++
++ /* machine stream operations */
++ struct snd_soc_ops *ops;
++
++ /* CPU <--> Codec DAI links */
++ struct snd_soc_dai_link *dai_link;
++ int num_links;
++};
++
++/* SoC Device - the audio subsystem */
++struct snd_soc_device {
++ struct device *dev;
++ struct snd_soc_machine *machine;
++ struct snd_soc_platform *platform;
++ struct snd_soc_codec *codec;
++ struct snd_soc_codec_device *codec_dev;
++ void *codec_data;
++};
++
++/* runtime channel data */
++struct snd_soc_pcm_runtime {
++ struct snd_soc_codec_dai *codec_dai;
++ struct snd_soc_cpu_dai *cpu_dai;
++ struct snd_soc_device *socdev;
++};
++
++/* enumerated kcontrol */
++struct soc_enum {
++ unsigned short reg;
++ unsigned short reg2;
++ unsigned char shift_l;
++ unsigned char shift_r;
++ unsigned int mask;
++ const char **texts;
++ void *dapm;
++};
++
++/* clocking configuration data */
++struct snd_soc_clock_info {
++ unsigned int rate;
++ unsigned int fs;
++ unsigned int bclk_master;
++};
++
++#endif
+Index: linux-2.6-pxa-new/sound/Kconfig
+===================================================================
+--- linux-2.6-pxa-new.orig/sound/Kconfig
++++ linux-2.6-pxa-new/sound/Kconfig
+@@ -76,6 +76,8 @@ source "sound/sparc/Kconfig"
+
+ source "sound/parisc/Kconfig"
+
++source "sound/soc/Kconfig"
++
+ endmenu
+
+ menu "Open Sound System"
+Index: linux-2.6-pxa-new/sound/soc/Kconfig
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/Kconfig
+@@ -0,0 +1,37 @@
++#
++# SoC audio configuration
++#
++
++menu "SoC audio support"
++ depends on SND!=n
++
++config SND_SOC_AC97_BUS
++ bool
++
++config SND_SOC
++ tristate "SoC audio support"
++ ---help---
++
++ If you want SoC support, you should say Y here and also to the
++ specific driver for your SoC below. You will also need to select the
++ specific codec(s) attached to the SoC
++
++ This SoC audio support can also be built as a module. If so, the module
++ will be called snd-soc-core.
++
++# All the supported Soc's
++menu "Soc Platforms"
++depends on SND_SOC
++source "sound/soc/pxa/Kconfig"
++source "sound/soc/at91/Kconfig"
++source "sound/soc/imx/Kconfig"
++source "sound/soc/s3c24xx/Kconfig"
++endmenu
++
++# Supported codecs
++menu "Soc Codecs"
++depends on SND_SOC
++source "sound/soc/codecs/Kconfig"
++endmenu
++
++endmenu
+Index: linux-2.6-pxa-new/sound/soc/Makefile
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/Makefile
+@@ -0,0 +1,4 @@
++snd-soc-core-objs := soc-core.o soc-dapm.o
++
++obj-$(CONFIG_SND_SOC) += snd-soc-core.o
++obj-$(CONFIG_SND_SOC) += pxa/ at91/ imx/ s3c24xx/ codecs/
+Index: linux-2.6-pxa-new/sound/soc/codecs/Kconfig
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/Kconfig
+@@ -0,0 +1,90 @@
++config SND_SOC_AC97_CODEC
++ tristate "SoC generic AC97 support"
++ depends SND_SOC
++ help
++ Say Y or M if you want generic AC97 support. This is not required
++ for the AC97 codecs listed below.
++
++config SND_SOC_WM8711
++ tristate "SoC driver for the WM8711 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8711 codec.
++
++config SND_SOC_WM8510
++ tristate "SoC driver for the WM8510 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8711 codec.
++
++config SND_SOC_WM8731
++ tristate "SoC driver for the WM8731 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8731 codec.
++
++config SND_SOC_WM8750
++ tristate "SoC driver for the WM8750 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8750 codec.
++
++config SND_SOC_WM8753
++ tristate "SoC driver for the WM8753 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8753 codec.
++
++config SND_SOC_WM8772
++ tristate "SoC driver for the WM8772 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8772 codec.
++
++config SND_SOC_WM8971
++ tristate "SoC driver for the WM8971 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8971 codec.
++
++config SND_SOC_WM8976
++ tristate "SoC driver for the WM8976 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8976 codec.
++
++config SND_SOC_WM8974
++ tristate "SoC driver for the WM8974 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8974 codec.
++
++config SND_SOC_WM8980
++ tristate "SoC driver for the WM8980 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM8980 codec.
++
++config SND_SOC_WM9713
++ tristate "SoC driver for the WM9713 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM9713 codec.
++
++config SND_SOC_WM9712
++ tristate "SoC driver for the WM9712 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the WM9712 codec.
++
++config SND_SOC_UDA1380
++ tristate "SoC driver for the UDA1380 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the UDA1380 codec.
++
++config SND_SOC_AK4535
++ tristate "SoC driver for the AK4535 codec"
++ depends SND_SOC
++ help
++ Say Y or M if you want to support the AK4535 codec.
+Index: linux-2.6-pxa-new/sound/soc/codecs/Makefile
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/Makefile
+@@ -0,0 +1,31 @@
++snd-soc-ac97-objs := ac97.o
++snd-soc-wm8711-objs := wm8711.o
++snd-soc-wm8510-objs := wm8510.o
++snd-soc-wm8731-objs := wm8731.o
++snd-soc-wm8750-objs := wm8750.o
++snd-soc-wm8753-objs := wm8753.o
++snd-soc-wm8772-objs := wm8772.o
++snd-soc-wm8971-objs := wm8971.o
++snd-soc-wm8974-objs := wm8974.o
++snd-soc-wm8976-objs := wm8976.o
++snd-soc-wm8980-objs := wm8980.o
++snd-soc-uda1380-objs := uda1380.o
++snd-soc-ak4535-objs := ak4535.o
++snd-soc-wm9713-objs := wm9713.o
++snd-soc-wm9712-objs := wm9712.o
++
++obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
++obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o
++obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
++obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
++obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
++obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
++obj-$(CONFIG_SND_SOC_WM8772) += snd-soc-wm8772.o
++obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
++obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o
++obj-$(CONFIG_SND_SOC_WM8976) += snd-soc-wm8976.o
++obj-$(CONFIG_SND_SOC_WM8980) += snd-soc-wm8980.o
++obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
++obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
++obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
++obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
+Index: linux-2.6-pxa-new/sound/soc/codecs/ac97.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/ac97.c
+@@ -0,0 +1,167 @@
++/*
++ * ac97.c -- ALSA Soc AC97 codec support
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 17th Oct 2005 Initial version.
++ *
++ * Generic AC97 support.
++ */
++
++#include <linux/init.h>
++#include <linux/kernel.h>
++#include <linux/device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/ac97_codec.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#define AC97_VERSION "0.5"
++
++#define AC97_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define AC97_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++/* may need to expand this */
++static struct snd_soc_dai_mode soc_ac97[] = {
++ {0, 0, SNDRV_PCM_FMTBIT_S16_LE, AC97_RATES},
++ {0, 0, SNDRV_PCM_FMTBIT_S18_3LE, AC97_RATES},
++ {0, 0, SNDRV_PCM_FMTBIT_S20_3LE, AC97_RATES},
++};
++
++static int ac97_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++
++ int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
++ AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
++ return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
++}
++
++static struct snd_soc_codec_dai ac97_dai = {
++ .name = "AC97 HiFi",
++ .playback = {
++ .stream_name = "AC97 Playback",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .stream_name = "AC97 Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .ops = {
++ .prepare = ac97_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(soc_ac97),
++ .mode = soc_ac97,},
++};
++
++static unsigned int ac97_read(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ return soc_ac97_ops.read(codec->ac97, reg);
++}
++
++static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int val)
++{
++ soc_ac97_ops.write(codec->ac97, reg, val);
++ return 0;
++}
++
++static int ac97_soc_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec;
++ struct snd_ac97_bus *ac97_bus;
++ struct snd_ac97_template ac97_template;
++ int ret = 0;
++
++ printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
++
++ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (socdev->codec == NULL)
++ return -ENOMEM;
++ codec = socdev->codec;
++ mutex_init(&codec->mutex);
++
++ codec->name = "AC97";
++ codec->owner = THIS_MODULE;
++ codec->dai = &ac97_dai;
++ codec->num_dai = 1;
++ codec->write = ac97_write;
++ codec->read = ac97_read;
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if(ret < 0)
++ goto err;
++
++ /* add codec as bus device for standard ac97 */
++ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus);
++ if(ret < 0)
++ goto bus_err;
++
++ memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
++ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97);
++ if(ret < 0)
++ goto bus_err;
++
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0)
++ goto bus_err;
++ return 0;
++
++bus_err:
++ snd_soc_free_pcms(socdev);
++
++err:
++ kfree(socdev->codec->reg_cache);
++ kfree(socdev->codec);
++ socdev->codec = NULL;
++ return ret;
++}
++
++static int ac97_soc_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if(codec == NULL)
++ return 0;
++
++ snd_soc_free_pcms(socdev);
++ kfree(socdev->codec->reg_cache);
++ kfree(socdev->codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_ac97= {
++ .probe = ac97_soc_probe,
++ .remove = ac97_soc_remove,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
++
++MODULE_DESCRIPTION("Soc Generic AC97 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/ac97.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/ac97.h
+@@ -0,0 +1,18 @@
++/*
++ * linux/sound/codecs/ac97.h -- ALSA SoC Layer
++ *
++ * Author: Liam Girdwood
++ * Created: Dec 1st 2005
++ * Copyright: Wolfson Microelectronics. PLC.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef __LINUX_SND_SOC_AC97_H
++#define __LINUX_SND_SOC_AC97_H
++
++extern struct snd_soc_codec_device soc_codec_dev_ac97;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/ak4535.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/ak4535.c
+@@ -0,0 +1,701 @@
++/*
++ * ak4535.c -- AK4535 ALSA Soc Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard@openedhand.com>
++ *
++ * Based on wm8753.c by Liam Girdwood
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "ak4535.h"
++
++#define AUDIO_NAME "ak4535"
++#define AK4535_VERSION "0.3"
++
++struct snd_soc_codec_device soc_codec_dev_ak4535;
++
++/*
++ * ak4535 register cache
++ */
++static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
++ 0x0000, 0x0080, 0x0000, 0x0003,
++ 0x0002, 0x0000, 0x0011, 0x0001,
++ 0x0000, 0x0040, 0x0036, 0x0010,
++ 0x0000, 0x0000, 0x0057, 0x0000,
++};
++
++#define AK4535_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | \
++ SND_SOC_DAIFMT_NB_NF)
++
++#define AK4535_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define AK4535_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++static struct snd_soc_dai_mode ak4535_modes[] = {
++ /* codec frame and clock slave modes */
++ {
++ .fmt = AK4535_DAIFMT,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = AK4535_RATES,
++ .pcmdir = AK4535_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++ {
++ .fmt = AK4535_DAIFMT,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = AK4535_RATES,
++ .pcmdir = AK4535_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 32,
++ },
++};
++
++/*
++ * read ak4535 register cache
++ */
++static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg >= AK4535_CACHEREGNUM)
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write ak4535 register cache
++ */
++static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
++ u16 reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg >= AK4535_CACHEREGNUM)
++ return;
++ cache[reg] = value;
++}
++
++/*
++ * write to the AK4535 register space
++ */
++static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D8 AK4535 register offset
++ * D7...D0 register data
++ */
++ data[0] = reg & 0xff;
++ data[1] = value & 0xff;
++
++ ak4535_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -EIO;
++}
++
++static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
++static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
++static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
++static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"};
++static const char *ak4535_mic_select[] = {"Internal", "External"};
++
++static const struct soc_enum ak4535_enum[] = {
++ SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain),
++ SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out),
++ SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out),
++ SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp),
++ SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select),
++};
++
++static const struct snd_kcontrol_new ak4535_snd_controls[] = {
++ SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0),
++ SOC_ENUM("Mono 1 Output", ak4535_enum[1]),
++ SOC_ENUM("Mono 1 Gain", ak4535_enum[0]),
++ SOC_ENUM("Headphone Output", ak4535_enum[2]),
++ SOC_ENUM("Playback Deemphasis", ak4535_enum[3]),
++ SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0),
++ SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0),
++ SOC_ENUM("Mic Select", ak4535_enum[4]),
++ SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0),
++ SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0),
++ SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0),
++ SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0),
++ SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0),
++ SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0),
++ SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0),
++ SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1),
++ SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1),
++ SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0),
++ SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
++};
++
++/* add non dapm controls */
++static int ak4535_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&ak4535_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ return 0;
++}
++
++/* Mono 1 Mixer */
++static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
++ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
++ SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0),
++};
++
++/* Stereo Mixer */
++static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = {
++ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0),
++ SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0),
++ SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0),
++};
++
++/* Input Mixer */
++static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = {
++ SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0),
++ SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0),
++};
++
++/* Input mux */
++static const struct snd_kcontrol_new ak4535_input_mux_control =
++ SOC_DAPM_ENUM("Input Select", ak4535_enum[0]);
++
++/* HP L switch */
++static const struct snd_kcontrol_new ak4535_hpl_control =
++ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1);
++
++/* HP R switch */
++static const struct snd_kcontrol_new ak4535_hpr_control =
++ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1);
++
++/* Speaker switch */
++static const struct snd_kcontrol_new ak4535_spk_control =
++ SOC_DAPM_SINGLE("Switch", AK4535_MODE2, 0, 0, 0);
++
++/* mono 2 switch */
++static const struct snd_kcontrol_new ak4535_mono2_control =
++ SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0);
++
++/* Line out switch */
++static const struct snd_kcontrol_new ak4535_line_control =
++ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0);
++
++/* ak4535 dapm widgets */
++static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = {
++ SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
++ &ak4535_stereo_mixer_controls[0],
++ ARRAY_SIZE(ak4535_stereo_mixer_controls)),
++ SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
++ &ak4535_mono1_mixer_controls[0],
++ ARRAY_SIZE(ak4535_mono1_mixer_controls)),
++ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
++ &ak4535_input_mixer_controls[0],
++ ARRAY_SIZE(ak4535_mono1_mixer_controls)),
++ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
++ &ak4535_input_mux_control),
++ SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0),
++ SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
++ &ak4535_mono2_control),
++ SND_SOC_DAPM_SWITCH("Speaker Enable", SND_SOC_NOPM, 0, 0,
++ &ak4535_spk_control),
++ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
++ &ak4535_line_control),
++ SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0,
++ &ak4535_hpl_control),
++ SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0,
++ &ak4535_hpr_control),
++ SND_SOC_DAPM_OUTPUT("LOUT"),
++ SND_SOC_DAPM_OUTPUT("HPL"),
++ SND_SOC_DAPM_OUTPUT("ROUT"),
++ SND_SOC_DAPM_OUTPUT("HPR"),
++ SND_SOC_DAPM_OUTPUT("SPP"),
++ SND_SOC_DAPM_OUTPUT("SPN"),
++ SND_SOC_DAPM_OUTPUT("MOUT1"),
++ SND_SOC_DAPM_OUTPUT("MOUT2"),
++ SND_SOC_DAPM_OUTPUT("MICOUT"),
++ SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 1),
++ SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0),
++
++ SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0),
++ SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0),
++ SND_SOC_DAPM_INPUT("MICIN"),
++ SND_SOC_DAPM_INPUT("MICEXT"),
++ SND_SOC_DAPM_INPUT("AUX"),
++ SND_SOC_DAPM_INPUT("MIN"),
++ SND_SOC_DAPM_INPUT("AIN"),
++};
++
++static const char *audio_map[][3] = {
++ /*stereo mixer */
++ {"Stereo Mixer", "Playback Switch", "DAC"},
++ {"Stereo Mixer", "Mic Sidetone Switch", "Mic"},
++ {"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
++
++ /* mono1 mixer */
++ {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"},
++ {"Mono1 Mixer", "Mono Playback Switch", "DAC"},
++
++ /* mono2 mixer */
++ {"Mono2 Mixer", "Mono Playback Switch", "Stereo Mixer"},
++
++ /* Mic */
++ {"AIN", NULL, "Mic"},
++ {"Input Mux", "Internal", "Mic Int Bias"},
++ {"Input Mux", "External", "Mic Ext Bias"},
++ {"Mic Int Bias", NULL, "MICIN"},
++ {"Mic Ext Bias", NULL, "MICEXT"},
++ {"MICOUT", NULL, "Input Mux"},
++
++ /* line out */
++ {"LOUT", "Switch", "Line"},
++ {"ROUT", "Switch", "Line Out Enable"},
++ {"Line Out Enable", NULL, "Line Out"},
++ {"Line Out", NULL, "Stereo Mixer"},
++
++ /* mono1 out */
++ {"MOUT1", NULL, "Mono Out"},
++ {"Mono Out", NULL, "Mono Mixer"},
++
++ /* left HP */
++ {"HPL", "Switch", "Left HP Enable"},
++ {"Left HP Enable", NULL, "HP L Amp"},
++ {"HP L Amp", NULL, "Stereo Mixer"},
++
++ /* right HP */
++ {"HPR", "Switch", "Right HP Enable"},
++ {"Right HP Enable", NULL, "HP R Amp"},
++ {"HP R Amp", NULL, "Stereo Mixer"},
++
++ /* speaker */
++ {"SPP", "Switch", "Speaker Enable"},
++ {"SPN", "Switch", "Speaker Enable"},
++ {"Speaker Enable", NULL, "Spk Amp"},
++ {"Spk Amp", NULL, "MIN"},
++
++ /* mono 2 */
++ {"MOUT2", "Switch", "Mono 2 Enable"},
++ {"Mono 2 Enable", NULL, "Stereo Mixer"},
++
++ /* Aux In */
++ {"Aux In", NULL, "AUX"},
++
++ /* ADC */
++ {"ADC", NULL, "Input Mixer"},
++ {"Input Mixer", "Mic Capture Switch", "Mic"},
++ {"Input Mixer", "Aux Capture Switch", "Aux In"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int ak4535_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(ak4535_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &ak4535_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++static int ak4535_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u8 mode = 0, mode2;
++ int bfs;
++
++ mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2);
++ bfs = SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs);
++ snd_assert(bfs, return -ENODEV);
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ mode = 0x0002;
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ mode = 0x0001;
++ break;
++ }
++
++ /* set fs */
++ switch (rtd->codec_dai->dai_runtime.fs) {
++ case 1024:
++ mode2 |= (0x3 << 5);
++ break;
++ case 512:
++ mode2 |= (0x2 << 5);
++ break;
++ case 256:
++ mode2 |= (0x1 << 5);
++ break;
++ }
++
++ /* bfs */
++ if (bfs == 64)
++ mode |= 0x4;
++
++ /* set rate */
++ ak4535_write(codec, AK4535_MODE1, mode);
++ ak4535_write(codec, AK4535_MODE2, mode2);
++
++ return 0;
++}
++
++static unsigned int ak4535_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ if (info->fs != 256)
++ return 0;
++
++ /* we only support 256 FS atm */
++ if (info->rate * info->fs == clk) {
++ dai->mclk = clk;
++ return clk;
++ }
++
++ return 0;
++}
++
++static int ak4535_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
++ if (mute)
++ ak4535_write(codec, AK4535_DAC, mute_reg);
++ else
++ ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
++ return 0;
++}
++
++static int ak4535_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* vref/mid, clk and osc on, dac unmute, active */
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except vref/vmid, dac mute, inactive */
++ ak4535_write(codec, AK4535_PM1, 0x80);
++ ak4535_write(codec, AK4535_PM2, 0x0);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* everything off, inactive */
++ ak4535_write(codec, AK4535_PM1, 0x0);
++ ak4535_write(codec, AK4535_PM2, 0x80);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai ak4535_dai = {
++ .name = "AK4535",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = ak4535_config_sysclk,
++ .digital_mute = ak4535_mute,
++ .ops = {
++ .prepare = ak4535_pcm_prepare,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(ak4535_modes),
++ .mode = ak4535_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(ak4535_dai);
++
++static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int ak4535_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(ak4535_reg); i++) {
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ ak4535_dapm_event(codec, codec->suspend_dapm_state);
++ return 0;
++}
++
++/*
++ * initialise the AK4535 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int ak4535_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int ret = 0;
++
++ codec->name = "AK4535";
++ codec->owner = THIS_MODULE;
++ codec->read = ak4535_read_reg_cache;
++ codec->write = ak4535_write;
++ codec->dapm_event = ak4535_dapm_event;
++ codec->dai = &ak4535_dai;
++ codec->num_dai = 1;
++ codec->reg_cache_size = ARRAY_SIZE(ak4535_reg);
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(ak4535_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, ak4535_reg,
++ sizeof(u16) * ARRAY_SIZE(ak4535_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ak4535_reg);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* power on device */
++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ ak4535_add_controls(codec);
++ ak4535_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++static struct snd_soc_device *ak4535_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */
++
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver ak4535_i2c_driver;
++static struct i2c_client client_template;
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = ak4535_socdev;
++ struct ak4535_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL){
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if (ret < 0) {
++ printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = ak4535_init(socdev);
++ if (ret < 0) {
++ printk(KERN_ERR "failed to initialise AK4535\n");
++ goto err;
++ }
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int ak4535_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec* codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++
++ return 0;
++}
++
++static int ak4535_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, ak4535_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver ak4535_i2c_driver = {
++ .driver = {
++ .name = "AK4535 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_AK4535,
++ .attach_adapter = ak4535_i2c_attach,
++ .detach_client = ak4535_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "AK4535",
++ .driver = &ak4535_i2c_driver,
++};
++#endif
++
++static int ak4535_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct ak4535_setup_data *setup;
++ struct snd_soc_codec* codec;
++ int ret = 0;
++
++ printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ ak4535_socdev = socdev;
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&ak4535_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++ return ret;
++}
++
++/* power down chip */
++static int ak4535_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec* codec = socdev->codec;
++
++ if (codec->control_data)
++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&ak4535_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_ak4535 = {
++ .probe = ak4535_probe,
++ .remove = ak4535_remove,
++ .suspend = ak4535_suspend,
++ .resume = ak4535_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
++
++MODULE_DESCRIPTION("Soc AK4535 driver");
++MODULE_AUTHOR("Richard Purdie");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/ak4535.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/ak4535.h
+@@ -0,0 +1,46 @@
++/*
++ * ak4535.h -- AK4535 Soc Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard@openedhand.com>
++ *
++ * Based on wm8753.h
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef _AK4535_H
++#define _AK4535_H
++
++/* AK4535 register space */
++
++#define AK4535_PM1 0x0
++#define AK4535_PM2 0x1
++#define AK4535_SIG1 0x2
++#define AK4535_SIG2 0x3
++#define AK4535_MODE1 0x4
++#define AK4535_MODE2 0x5
++#define AK4535_DAC 0x6
++#define AK4535_MIC 0x7
++#define AK4535_TIMER 0x8
++#define AK4535_ALC1 0x9
++#define AK4535_ALC2 0xa
++#define AK4535_PGA 0xb
++#define AK4535_LATT 0xc
++#define AK4535_RATT 0xd
++#define AK4535_VOL 0xe
++#define AK4535_STATUS 0xf
++
++#define AK4535_CACHEREGNUM 0x10
++
++struct ak4535_setup_data {
++ unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai ak4535_dai;
++extern struct snd_soc_codec_device soc_codec_dev_ak4535;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/uda1380.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/uda1380.c
+@@ -0,0 +1,582 @@
++/*
++ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
++ * codec model.
++ *
++ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
++ * Copyright 2005 Openedhand Ltd.
++ */
++
++#include <linux/module.h>
++#include <linux/init.h>
++#include <linux/types.h>
++#include <linux/string.h>
++#include <linux/slab.h>
++#include <linux/errno.h>
++#include <linux/ioctl.h>
++#include <linux/delay.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/control.h>
++#include <sound/initval.h>
++#include <sound/info.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include "uda1380.h"
++
++#define UDA1380_VERSION "0.4"
++
++/*
++ * uda1380 register cache
++ */
++static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
++ 0x0502, 0x0000, 0x0000, 0x3f3f,
++ 0x0202, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0xff00, 0x0000, 0x4800,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x8000, 0x0002, 0x0000,
++};
++
++#define UDA1380_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | \
++ SND_SOC_DAIFMT_NB_NF)
++
++#define UDA1380_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define UDA1380_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++static struct snd_soc_dai_mode uda1380_modes[] = {
++ /* slave rates capture & playback */
++ {
++ .fmt = UDA1380_DAIFMT,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = UDA1380_RATES,
++ .pcmdir = UDA1380_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++
++ /* slave rates playback */
++ {
++ .fmt = UDA1380_DAIFMT,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++};
++
++/*
++ * read uda1380 register cache
++ */
++static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg == UDA1380_RESET)
++ return 0;
++ if (reg >= UDA1380_CACHEREGNUM)
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write uda1380 register cache
++ */
++static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
++ u16 reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg >= UDA1380_CACHEREGNUM)
++ return;
++ cache[reg] = value;
++}
++
++/*
++ * write to the UDA1380 register space
++ */
++static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[3];
++
++ /* data is
++ * data[0] is register offset
++ * data[1] is MS byte
++ * data[2] is LS byte
++ */
++ data[0] = reg;
++ data[1] = (value & 0xff00) >> 8;
++ data[2] = value & 0x00ff;
++
++ uda1380_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 3) == 3)
++ return 0;
++ else
++ return -EIO;
++}
++
++#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
++
++/* declarations of ALSA reg_elem_REAL controls */
++static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz",
++ "96kHz"};
++static const char *uda1380_input_sel[] = {"Line", "Mic"};
++
++static const struct soc_enum uda1380_enum[] = {
++ SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp),
++ SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel),
++};
++
++static const struct snd_kcontrol_new uda1380_snd_controls[] = {
++ SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 127, 0),
++ SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0),
++ SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0),
++ SOC_ENUM("Playback De-emphasis", uda1380_enum[0]),
++ SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0),
++ SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0),
++ SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0),
++ SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 0),
++ SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0),
++ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0),
++ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1),
++ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
++};
++
++/* add non dapm controls */
++static int uda1380_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ return 0;
++}
++
++/* Input mux */
++static const struct snd_kcontrol_new uda1380_input_mux_control =
++ SOC_DAPM_ENUM("Input Select", uda1380_enum[1]);
++
++static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
++ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
++ &uda1380_input_mux_control),
++ SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
++ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
++ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
++ SND_SOC_DAPM_INPUT("VINM"),
++ SND_SOC_DAPM_INPUT("VINL"),
++ SND_SOC_DAPM_INPUT("VINR"),
++ SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
++ SND_SOC_DAPM_OUTPUT("VOUTLHP"),
++ SND_SOC_DAPM_OUTPUT("VOUTRHP"),
++ SND_SOC_DAPM_OUTPUT("VOUTL"),
++ SND_SOC_DAPM_OUTPUT("VOUTR"),
++ SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
++ SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
++};
++
++static const char *audio_map[][3] = {
++
++ /* analog mixer setup is different from diagram for dapm */
++ {"HeadPhone Driver", NULL, "Analog Mixer"},
++ {"VOUTR", NULL, "Analog Mixer"},
++ {"VOUTL", NULL, "Analog Mixer"},
++ {"Analog Mixer", NULL, "VINR"},
++ {"Analog Mixer", NULL, "VINL"},
++ {"Analog Mixer", NULL, "DAC"},
++
++ /* headphone driver */
++ {"VOUTLHP", NULL, "HeadPhone Driver"},
++ {"VOUTRHP", NULL, "HeadPhone Driver"},
++
++ /* input mux */
++ {"Left ADC", NULL, "Input Mux"},
++ {"Input Mux", "Mic", "Mic LNA"},
++ {"Input Mux", "Line", "Left PGA"},
++
++ /* right input */
++ {"Right ADC", NULL, "Right PGA"},
++
++ /* inputs */
++ {"Mic LNA", NULL, "VINM"},
++ {"Left PGA", NULL, "VINL"},
++ {"Right PGA", NULL, "VINR"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int uda1380_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ uda1380_write(codec, UDA1380_CLK, R00_EN_DAC | R00_EN_INT | clk);
++ else
++ uda1380_write(codec, UDA1380_CLK, R00_EN_ADC | R00_EN_DEC | clk);
++
++ return 0;
++}
++
++static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ uda1380_write(codec, UDA1380_CLK, ~(R00_EN_DAC | R00_EN_INT) & clk);
++ else
++ uda1380_write(codec, UDA1380_CLK, ~(R00_EN_ADC | R00_EN_DEC) & clk);
++}
++
++static unsigned int uda1380_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ if(info->fs != 256)
++ return 0;
++
++ /* we only support 256 FS atm */
++ if(info->rate * info->fs == clk) {
++ dai->mclk = clk;
++ return clk;
++ }
++
++ return 0;
++}
++
++static int uda1380_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff;
++ if(mute)
++ uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000);
++ else
++ uda1380_write(codec, UDA1380_DEEMP, mute_reg);
++ return 0;
++}
++
++static int uda1380_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except internal bias */
++ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* everything off, inactive */
++ uda1380_write(codec, UDA1380_PM, 0x0);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai uda1380_dai = {
++ .name = "UDA1380",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = uda1380_config_sysclk,
++ .digital_mute = uda1380_mute,
++ .ops = {
++ .prepare = uda1380_pcm_prepare,
++ .shutdown = uda1380_pcm_shutdown,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(uda1380_modes),
++ .mode = uda1380_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(uda1380_dai);
++
++static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int uda1380_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ uda1380_dapm_event(codec, codec->suspend_dapm_state);
++ return 0;
++}
++
++/*
++ * initialise the UDA1380 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int uda1380_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int ret = 0;
++
++ codec->name = "UDA1380";
++ codec->owner = THIS_MODULE;
++ codec->read = uda1380_read_reg_cache;
++ codec->write = uda1380_write;
++ codec->dapm_event = uda1380_dapm_event;
++ codec->dai = &uda1380_dai;
++ codec->num_dai = 1;
++ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(uda1380_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, uda1380_reg,
++ sizeof(u16) * ARRAY_SIZE(uda1380_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(uda1380_reg);
++ uda1380_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if(ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* power on device */
++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ uda1380_write(codec, UDA1380_CLK, 0);
++
++ /* uda1380 init */
++ uda1380_add_controls(codec);
++ uda1380_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if(ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++static struct snd_soc_device *uda1380_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */
++
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver uda1380_i2c_driver;
++static struct i2c_client client_template;
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++
++static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = uda1380_socdev;
++ struct uda1380_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL){
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if(ret < 0) {
++ printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = uda1380_init(socdev);
++ if(ret < 0) {
++ printk(KERN_ERR "failed to initialise UDA1380\n");
++ goto err;
++ }
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int uda1380_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec* codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int uda1380_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, uda1380_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver uda1380_i2c_driver = {
++ .driver = {
++ .name = "UDA1380 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_UDA1380,
++ .attach_adapter = uda1380_i2c_attach,
++ .detach_client = uda1380_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "UDA1380",
++ .driver = &uda1380_i2c_driver,
++};
++#endif
++
++static int uda1380_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct uda1380_setup_data *setup;
++ struct snd_soc_codec* codec;
++ int ret = 0;
++
++ printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ uda1380_socdev = socdev;
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&uda1380_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++ return ret;
++}
++
++/* power down chip */
++static int uda1380_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec* codec = socdev->codec;
++
++ if (codec->control_data)
++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&uda1380_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_uda1380 = {
++ .probe = uda1380_probe,
++ .remove = uda1380_remove,
++ .suspend = uda1380_suspend,
++ .resume = uda1380_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
++
++MODULE_AUTHOR("Giorgio Padrin");
++MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/uda1380.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/uda1380.h
+@@ -0,0 +1,56 @@
++/*
++ * Audio support for Philips UDA1380
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
++ */
++
++#define UDA1380_CLK 0x00
++#define UDA1380_IFACE 0x01
++#define UDA1380_PM 0x02
++#define UDA1380_AMIX 0x03
++#define UDA1380_HP 0x04
++#define UDA1380_MVOL 0x10
++#define UDA1380_MIXVOL 0x11
++#define UDA1380_MODE 0x12
++#define UDA1380_DEEMP 0x13
++#define UDA1380_MIXER 0x14
++#define UDA1380_INTSTAT 0x18
++#define UDA1380_DEC 0x20
++#define UDA1380_PGA 0x21
++#define UDA1380_ADC 0x22
++#define UDA1380_AGC 0x23
++#define UDA1380_DECSTAT 0x28
++#define UDA1380_RESET 0x7f
++
++#define UDA1380_CACHEREGNUM 0x24
++
++/* Register flags */
++#define R00_EN_ADC 0x0800
++#define R00_EN_DEC 0x0400
++#define R00_EN_DAC 0x0200
++#define R00_EN_INT 0x0100
++#define R02_PON_HP 0x2000
++#define R02_PON_DAC 0x0400
++#define R02_PON_BIAS 0x0100
++#define R02_PON_LNA 0x0010
++#define R02_PON_PGAL 0x0008
++#define R02_PON_ADCL 0x0004
++#define R02_PON_PGAR 0x0002
++#define R02_PON_ADCR 0x0001
++#define R13_MTM 0x4000
++#define R21_MT_ADC 0x8000
++#define R22_SEL_LNA 0x0008
++#define R22_SEL_MIC 0x0004
++#define R22_SKIP_DCFIL 0x0002
++#define R23_AGC_EN 0x0001
++
++struct uda1380_setup_data {
++ unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai uda1380_dai;
++extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8731.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8731.c
+@@ -0,0 +1,886 @@
++/*
++ * wm8731.c -- WM8731 ALSA SoC Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard@openedhand.com>
++ *
++ * Based on wm8753.c by Liam Girdwood
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8731.h"
++
++#define AUDIO_NAME "wm8731"
++#define WM8731_VERSION "0.12"
++
++/*
++ * Debug
++ */
++
++#define WM8731_DEBUG 0
++
++#ifdef WM8731_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++struct snd_soc_codec_device soc_codec_dev_wm8731;
++
++/*
++ * wm8731 register cache
++ * We can't read the WM8731 register space when we are
++ * using 2 wire for device control, so we cache them instead.
++ * There is no point in caching the reset register
++ */
++static const u16 wm8731_reg[WM8731_CACHEREGNUM] = {
++ 0x0097, 0x0097, 0x0079, 0x0079,
++ 0x000a, 0x0008, 0x009f, 0x000a,
++ 0x0000, 0x0000
++};
++
++#define WM8731_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
++ SND_SOC_DAIFMT_IB_IF)
++
++#define WM8731_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM8731_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8731_HIFI_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++static struct snd_soc_dai_mode wm8731_modes[] = {
++ /* codec frame and clock master modes */
++ /* 8k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 1536,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 2304,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 1408,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 2112,
++ .bfs = 64,
++ },
++
++ /* 32k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 384,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 576,
++ .bfs = 64,
++ },
++
++ /* 44.1k & 48k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 384,
++ .bfs = 64,
++ },
++
++ /* 88.2 & 96k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 128,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 192,
++ .bfs = 64,
++ },
++
++ /* USB codec frame and clock master modes */
++ /* 8k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1500,
++ .bfs = SND_SOC_FSBD(1),
++ },
++
++ /* 44.1k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 272,
++ .bfs = SND_SOC_FSBD(1),
++ },
++
++ /* 48k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 250,
++ .bfs = SND_SOC_FSBD(1),
++ },
++
++ /* 88.2k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 136,
++ .bfs = SND_SOC_FSBD(1),
++ },
++
++ /* 96k */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 125,
++ .bfs = SND_SOC_FSBD(1),
++ },
++
++ /* codec frame and clock slave modes */
++ {
++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8731_HIFI_BITS,
++ .pcmrate = WM8731_RATES,
++ .pcmdir = WM8731_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++/*
++ * read wm8731 register cache
++ */
++static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg == WM8731_RESET)
++ return 0;
++ if (reg >= WM8731_CACHEREGNUM)
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write wm8731 register cache
++ */
++static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec,
++ u16 reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg >= WM8731_CACHEREGNUM)
++ return;
++ cache[reg] = value;
++}
++
++/*
++ * write to the WM8731 register space
++ */
++static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D9 WM8731 register offset
++ * D8...D0 register data
++ */
++ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++ data[1] = value & 0x00ff;
++
++ wm8731_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -EIO;
++}
++
++#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0)
++
++static const char *wm8731_input_select[] = {"Line In", "Mic"};
++static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
++
++static const struct soc_enum wm8731_enum[] = {
++ SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select),
++ SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph),
++};
++
++static const struct snd_kcontrol_new wm8731_snd_controls[] = {
++
++SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V,
++ 0, 127, 0),
++SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V,
++ 7, 1, 0),
++
++SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0),
++SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1),
++
++SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0),
++SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1),
++
++SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1),
++
++SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1),
++SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
++
++SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
++};
++
++/* add non dapm controls */
++static int wm8731_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
++ if ((err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0)
++ return err;
++ }
++
++ return 0;
++}
++
++/* Output Mixer */
++static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
++SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
++SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
++};
++
++/* Input mux */
++static const struct snd_kcontrol_new wm8731_input_mux_controls =
++SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
++
++static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
++SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
++ &wm8731_output_mixer_controls[0],
++ ARRAY_SIZE(wm8731_output_mixer_controls)),
++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1),
++SND_SOC_DAPM_OUTPUT("LOUT"),
++SND_SOC_DAPM_OUTPUT("LHPOUT"),
++SND_SOC_DAPM_OUTPUT("ROUT"),
++SND_SOC_DAPM_OUTPUT("RHPOUT"),
++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1),
++SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &wm8731_input_mux_controls),
++SND_SOC_DAPM_PGA("Line Input", WM8731_PWR, 0, 1, NULL, 0),
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8731_PWR, 1, 1),
++SND_SOC_DAPM_INPUT("MICIN"),
++SND_SOC_DAPM_INPUT("RLINEIN"),
++SND_SOC_DAPM_INPUT("LLINEIN"),
++};
++
++static const char *intercon[][3] = {
++ /* output mixer */
++ {"Output Mixer", "Line Bypass Switch", "Line Input"},
++ {"Output Mixer", "HiFi Playback Switch", "DAC"},
++ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
++
++ /* outputs */
++ {"RHPOUT", NULL, "Output Mixer"},
++ {"ROUT", NULL, "Output Mixer"},
++ {"LHPOUT", NULL, "Output Mixer"},
++ {"LOUT", NULL, "Output Mixer"},
++
++ /* input mux */
++ {"Input Mux", "Line In", "Line Input"},
++ {"Input Mux", "Mic", "Mic Bias"},
++ {"ADC", NULL, "Input Mux"},
++
++ /* inputs */
++ {"Line Input", NULL, "LLINEIN"},
++ {"Line Input", NULL, "RLINEIN"},
++ {"Mic Bias", NULL, "MICIN"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int wm8731_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
++ }
++
++ /* set up audio path interconnects */
++ for(i = 0; intercon[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, intercon[i][0],
++ intercon[i][1], intercon[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++struct _coeff_div {
++ u32 mclk;
++ u32 rate;
++ u16 fs;
++ u8 sr:4;
++ u8 bosr:1;
++ u8 usb:1;
++};
++
++/* codec mclk clock divider coefficients */
++static const struct _coeff_div coeff_div[] = {
++ /* 48k */
++ {12288000, 48000, 256, 0x0, 0x0, 0x0},
++ {18432000, 48000, 384, 0x0, 0x1, 0x0},
++ {12000000, 48000, 250, 0x0, 0x0, 0x1},
++
++ /* 32k */
++ {12288000, 32000, 384, 0x6, 0x0, 0x0},
++ {18432000, 32000, 576, 0x6, 0x1, 0x0},
++
++ /* 8k */
++ {12288000, 8000, 1536, 0x3, 0x0, 0x0},
++ {18432000, 8000, 2304, 0x3, 0x1, 0x0},
++ {11289600, 8000, 1408, 0xb, 0x0, 0x0},
++ {16934400, 8000, 2112, 0xb, 0x1, 0x0},
++ {12000000, 8000, 1500, 0x3, 0x0, 0x1},
++
++ /* 96k */
++ {12288000, 96000, 128, 0x7, 0x0, 0x0},
++ {18432000, 96000, 192, 0x7, 0x1, 0x0},
++ {12000000, 96000, 125, 0x7, 0x0, 0x1},
++
++ /* 44.1k */
++ {11289600, 44100, 256, 0x8, 0x0, 0x0},
++ {16934400, 44100, 384, 0x8, 0x1, 0x0},
++ {12000000, 44100, 272, 0x8, 0x1, 0x1},
++
++ /* 88.2k */
++ {11289600, 88200, 128, 0xf, 0x0, 0x0},
++ {16934400, 88200, 192, 0xf, 0x1, 0x0},
++ {12000000, 88200, 136, 0xf, 0x1, 0x1},
++};
++
++static inline int get_coeff(int mclk, int rate)
++{
++ int i;
++
++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
++ return i;
++ }
++ return 0;
++}
++
++/* WM8731 supports numerous clocks per sample rate */
++static unsigned int wm8731_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ dai->mclk = 0;
++
++ /* check that the calculated FS and rate actually match a clock from
++ * the machine driver */
++ if (info->fs * info->rate == clk)
++ dai->mclk = clk;
++
++ return dai->mclk;
++}
++
++static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 iface = 0, srate;
++ int i = get_coeff(rtd->codec_dai->mclk,
++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate));
++
++ /* set master/slave audio interface */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ iface |= 0x0040;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ break;
++ }
++ srate = (coeff_div[i].sr << 2) |
++ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
++ wm8731_write(codec, WM8731_SRATE, srate);
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ iface |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ iface |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ iface |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ iface |= 0x0013;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ iface |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ iface |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ iface |= 0x000c;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ break;
++ case SND_SOC_DAIFMT_IB_IF:
++ iface |= 0x0090;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ iface |= 0x0080;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ iface |= 0x0010;
++ break;
++ }
++
++ /* set iface */
++ wm8731_write(codec, WM8731_IFACE, iface);
++
++ /* set active */
++ wm8731_write(codec, WM8731_ACTIVE, 0x0001);
++ return 0;
++}
++
++static void wm8731_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++
++ /* deactivate */
++ if (!codec->active) {
++ udelay(50);
++ wm8731_write(codec, WM8731_ACTIVE, 0x0);
++ }
++}
++
++static int wm8731_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
++ if (mute)
++ wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8);
++ else
++ wm8731_write(codec, WM8731_APDIGI, mute_reg);
++ return 0;
++}
++
++static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* vref/mid, osc on, dac unmute */
++ wm8731_write(codec, WM8731_PWR, reg);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except vref/vmid, */
++ wm8731_write(codec, WM8731_PWR, reg | 0x0040);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* everything off, dac mute, inactive */
++ wm8731_write(codec, WM8731_ACTIVE, 0x0);
++ wm8731_write(codec, WM8731_PWR, 0xffff);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai wm8731_dai = {
++ .name = "WM8731",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = wm8731_config_sysclk,
++ .digital_mute = wm8731_mute,
++ .ops = {
++ .prepare = wm8731_pcm_prepare,
++ .shutdown = wm8731_shutdown,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8731_modes),
++ .mode = wm8731_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(wm8731_dai);
++
++static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm8731_write(codec, WM8731_ACTIVE, 0x0);
++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm8731_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ wm8731_dapm_event(codec, codec->suspend_dapm_state);
++ return 0;
++}
++
++/*
++ * initialise the WM8731 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8731_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg, ret = 0;
++
++ codec->name = "WM8731";
++ codec->owner = THIS_MODULE;
++ codec->read = wm8731_read_reg_cache;
++ codec->write = wm8731_write;
++ codec->dapm_event = wm8731_dapm_event;
++ codec->dai = &wm8731_dai;
++ codec->num_dai = 1;
++ codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
++
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8731_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache,
++ wm8731_reg, sizeof(u16) * ARRAY_SIZE(wm8731_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8731_reg);
++
++ wm8731_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* power on device */
++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* set the update bits */
++ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
++ wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100);
++ reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
++ wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100);
++ reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
++ wm8731_write(codec, WM8731_LINVOL, reg | 0x0100);
++ reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
++ wm8731_write(codec, WM8731_RINVOL, reg | 0x0100);
++
++ wm8731_add_controls(codec);
++ wm8731_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++static struct snd_soc_device *wm8731_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8731 2 wire address is determined by GPIO5
++ * state during powerup.
++ * low = 0x1a
++ * high = 0x1b
++ */
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver wm8731_i2c_driver;
++static struct i2c_client client_template;
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++
++static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = wm8731_socdev;
++ struct wm8731_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if (ret < 0) {
++ err("failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = wm8731_init(socdev);
++ if (ret < 0) {
++ err("failed to initialise WM8731\n");
++ goto err;
++ }
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int wm8731_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec* codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int wm8731_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, wm8731_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8731_i2c_driver = {
++ .driver = {
++ .name = "WM8731 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_WM8731,
++ .attach_adapter = wm8731_i2c_attach,
++ .detach_client = wm8731_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "WM8731",
++ .driver = &wm8731_i2c_driver,
++};
++#endif
++
++static int wm8731_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct wm8731_setup_data *setup;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ info("WM8731 Audio Codec %s", WM8731_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ wm8731_socdev = socdev;
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&wm8731_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++ return ret;
++}
++
++/* power down chip */
++static int wm8731_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec->control_data)
++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&wm8731_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8731 = {
++ .probe = wm8731_probe,
++ .remove = wm8731_remove,
++ .suspend = wm8731_suspend,
++ .resume = wm8731_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
++
++MODULE_DESCRIPTION("ASoC WM8731 driver");
++MODULE_AUTHOR("Richard Purdie");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8731.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8731.h
+@@ -0,0 +1,41 @@
++/*
++ * wm8731.h -- WM8731 Soc Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard@openedhand.com>
++ *
++ * Based on wm8753.h
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef _WM8731_H
++#define _WM8731_H
++
++/* WM8731 register space */
++
++#define WM8731_LINVOL 0x00
++#define WM8731_RINVOL 0x01
++#define WM8731_LOUT1V 0x02
++#define WM8731_ROUT1V 0x03
++#define WM8731_APANA 0x04
++#define WM8731_APDIGI 0x05
++#define WM8731_PWR 0x06
++#define WM8731_IFACE 0x07
++#define WM8731_SRATE 0x08
++#define WM8731_ACTIVE 0x09
++#define WM8731_RESET 0x0f
++
++#define WM8731_CACHEREGNUM 10
++
++struct wm8731_setup_data {
++ unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8731_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8731;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8750.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8750.c
+@@ -0,0 +1,1282 @@
++/*
++ * wm8750.c -- WM8750 ALSA SoC audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard@openedhand.com>
++ *
++ * Based on WM8753.c
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8750.h"
++
++#define AUDIO_NAME "WM8750"
++#define WM8750_VERSION "0.11"
++
++/*
++ * Debug
++ */
++
++#define WM8750_DEBUG 0
++
++#ifdef WM8750_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++static struct workqueue_struct *wm8750_workq = NULL;
++static struct work_struct wm8750_dapm_work;
++
++/*
++ * wm8750 register cache
++ * We can't read the WM8750 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8750_reg[] = {
++ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
++ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
++ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
++ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
++ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
++ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
++ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
++ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
++ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
++ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
++ 0x0079, 0x0079, 0x0079, /* 40 */
++};
++
++#define WM8750_HIFI_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
++ SND_SOC_DAIFMT_IB_IF)
++
++#define WM8750_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM8750_HIFI_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++#define WM8750_HIFI_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8750_HIFI_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++static struct snd_soc_dai_mode wm8750_modes[] = {
++ /* common codec frame and clock master modes */
++ /* 8k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1408,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 2304,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 2112,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1500,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* 11.025k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1024,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1088,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* 16k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 768,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1152,
++ .bfs = WM8750_HIFI_FSB
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 750,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* 22.05k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 512,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 768,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 544,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* 32k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 576,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 375,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* 44.1k & 48k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 272,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 250,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* 88.2k & 96k */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 128,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 192,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 136,
++ .bfs = WM8750_HIFI_FSB,
++ },
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 125,
++ .bfs = WM8750_HIFI_FSB,
++ },
++
++ /* codec frame and clock slave modes */
++ {
++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8750_HIFI_BITS,
++ .pcmrate = WM8750_HIFI_RATES,
++ .pcmdir = WM8750_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++/*
++ * read wm8750 register cache
++ */
++static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg > WM8750_CACHE_REGNUM)
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write wm8750 register cache
++ */
++static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg > WM8750_CACHE_REGNUM)
++ return;
++ cache[reg] = value;
++}
++
++static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D9 WM8753 register offset
++ * D8...D0 register data
++ */
++ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++ data[1] = value & 0x00ff;
++
++ wm8750_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -EIO;
++}
++
++#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0)
++
++/*
++ * WM8750 Controls
++ */
++static const char *wm8750_bass[] = {"Linear Control", "Adaptive Boost"};
++static const char *wm8750_bass_filter[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
++static const char *wm8750_treble[] = {"8kHz", "4kHz"};
++static const char *wm8750_3d_lc[] = {"200Hz", "500Hz"};
++static const char *wm8750_3d_uc[] = {"2.2kHz", "1.5kHz"};
++static const char *wm8750_3d_func[] = {"Capture", "Playback"};
++static const char *wm8750_alc_func[] = {"Off", "Right", "Left", "Stereo"};
++static const char *wm8750_ng_type[] = {"Constant PGA Gain",
++ "Mute ADC Output"};
++static const char *wm8750_line_mux[] = {"Line 1", "Line 2", "Line 3", "PGA",
++ "Differential"};
++static const char *wm8750_pga_sel[] = {"Line 1", "Line 2", "Line 3",
++ "Differential"};
++static const char *wm8750_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut",
++ "ROUT1"};
++static const char *wm8750_diff_sel[] = {"Line 1", "Line 2"};
++static const char *wm8750_adcpol[] = {"Normal", "L Invert", "R Invert",
++ "L + R Invert"};
++static const char *wm8750_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
++static const char *wm8750_mono_mux[] = {"Stereo", "Mono (Left)",
++ "Mono (Right)", "Digital Mono"};
++
++static const struct soc_enum wm8750_enum[] = {
++SOC_ENUM_SINGLE(WM8750_BASS, 7, 2, wm8750_bass),
++SOC_ENUM_SINGLE(WM8750_BASS, 6, 2, wm8750_bass_filter),
++SOC_ENUM_SINGLE(WM8750_TREBLE, 6, 2, wm8750_treble),
++SOC_ENUM_SINGLE(WM8750_3D, 5, 2, wm8750_3d_lc),
++SOC_ENUM_SINGLE(WM8750_3D, 6, 2, wm8750_3d_uc),
++SOC_ENUM_SINGLE(WM8750_3D, 7, 2, wm8750_3d_func),
++SOC_ENUM_SINGLE(WM8750_ALC1, 7, 4, wm8750_alc_func),
++SOC_ENUM_SINGLE(WM8750_NGATE, 1, 2, wm8750_ng_type),
++SOC_ENUM_SINGLE(WM8750_LOUTM1, 0, 5, wm8750_line_mux),
++SOC_ENUM_SINGLE(WM8750_ROUTM1, 0, 5, wm8750_line_mux),
++SOC_ENUM_SINGLE(WM8750_LADCIN, 6, 4, wm8750_pga_sel), /* 10 */
++SOC_ENUM_SINGLE(WM8750_RADCIN, 6, 4, wm8750_pga_sel),
++SOC_ENUM_SINGLE(WM8750_ADCTL2, 7, 4, wm8750_out3),
++SOC_ENUM_SINGLE(WM8750_ADCIN, 8, 2, wm8750_diff_sel),
++SOC_ENUM_SINGLE(WM8750_ADCDAC, 5, 4, wm8750_adcpol),
++SOC_ENUM_SINGLE(WM8750_ADCDAC, 1, 4, wm8750_deemph),
++SOC_ENUM_SINGLE(WM8750_ADCIN, 6, 4, wm8750_mono_mux), /* 16 */
++
++};
++
++static const struct snd_kcontrol_new wm8750_snd_controls[] = {
++
++SOC_DOUBLE_R("Capture Volume", WM8750_LINVOL, WM8750_RINVOL, 0, 63, 0),
++SOC_DOUBLE_R("Capture ZC Switch", WM8750_LINVOL, WM8750_RINVOL, 6, 1, 0),
++SOC_DOUBLE_R("Capture Switch", WM8750_LINVOL, WM8750_RINVOL, 7, 1, 1),
++
++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8750_LOUT1V,
++ WM8750_ROUT1V, 7, 1, 0),
++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8750_LOUT2V,
++ WM8750_ROUT2V, 7, 1, 0),
++
++SOC_ENUM("Playback De-emphasis", wm8750_enum[15]),
++
++SOC_ENUM("Capture Polarity", wm8750_enum[14]),
++SOC_SINGLE("Playback 6dB Attenuate", WM8750_ADCDAC, 7, 1, 0),
++SOC_SINGLE("Capture 6dB Attenuate", WM8750_ADCDAC, 8, 1, 0),
++
++SOC_DOUBLE_R("PCM Volume", WM8750_LDAC, WM8750_RDAC, 0, 255, 0),
++
++SOC_ENUM("Bass Boost", wm8750_enum[0]),
++SOC_ENUM("Bass Filter", wm8750_enum[1]),
++SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1),
++
++SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0),
++SOC_ENUM("Treble Cut-off", wm8750_enum[2]),
++
++SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0),
++SOC_SINGLE("3D Volume", WM8750_3D, 1, 15, 0),
++SOC_ENUM("3D Lower Cut-off", wm8750_enum[3]),
++SOC_ENUM("3D Upper Cut-off", wm8750_enum[4]),
++SOC_ENUM("3D Mode", wm8750_enum[5]),
++
++SOC_SINGLE("ALC Capture Target Volume", WM8750_ALC1, 0, 7, 0),
++SOC_SINGLE("ALC Capture Max Volume", WM8750_ALC1, 4, 7, 0),
++SOC_ENUM("ALC Capture Function", wm8750_enum[6]),
++SOC_SINGLE("ALC Capture ZC Switch", WM8750_ALC2, 7, 1, 0),
++SOC_SINGLE("ALC Capture Hold Time", WM8750_ALC2, 0, 15, 0),
++SOC_SINGLE("ALC Capture Decay Time", WM8750_ALC3, 4, 15, 0),
++SOC_SINGLE("ALC Capture Attack Time", WM8750_ALC3, 0, 15, 0),
++SOC_SINGLE("ALC Capture NG Threshold", WM8750_NGATE, 3, 31, 0),
++SOC_ENUM("ALC Capture NG Type", wm8750_enum[4]),
++SOC_SINGLE("ALC Capture NG Switch", WM8750_NGATE, 0, 1, 0),
++
++SOC_SINGLE("Left ADC Capture Volume", WM8750_LADC, 0, 255, 0),
++SOC_SINGLE("Right ADC Capture Volume", WM8750_RADC, 0, 255, 0),
++
++SOC_SINGLE("ZC Timeout Switch", WM8750_ADCTL1, 0, 1, 0),
++SOC_SINGLE("Playback Invert Switch", WM8750_ADCTL1, 1, 1, 0),
++
++SOC_SINGLE("Right Speaker Playback Invert Switch", WM8750_ADCTL2, 4, 1, 0),
++
++/* Unimplemented */
++/* ADCDAC Bit 0 - ADCHPD */
++/* ADCDAC Bit 4 - HPOR */
++/* ADCTL1 Bit 2,3 - DATSEL */
++/* ADCTL1 Bit 4,5 - DMONOMIX */
++/* ADCTL1 Bit 6,7 - VSEL */
++/* ADCTL2 Bit 2 - LRCM */
++/* ADCTL2 Bit 3 - TRI */
++/* ADCTL3 Bit 5 - HPFLREN */
++/* ADCTL3 Bit 6 - VROI */
++/* ADCTL3 Bit 7,8 - ADCLRM */
++/* ADCIN Bit 4 - LDCM */
++/* ADCIN Bit 5 - RDCM */
++
++SOC_DOUBLE_R("Mic Boost", WM8750_LADCIN, WM8750_RADCIN, 4, 3, 0),
++
++SOC_DOUBLE_R("Bypass Left Playback Volume", WM8750_LOUTM1,
++ WM8750_LOUTM2, 4, 7, 1),
++SOC_DOUBLE_R("Bypass Right Playback Volume", WM8750_ROUTM1,
++ WM8750_ROUTM2, 4, 7, 1),
++SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8750_MOUTM1,
++ WM8750_MOUTM2, 4, 7, 1),
++
++SOC_SINGLE("Mono Playback ZC Switch", WM8750_MOUTV, 7, 1, 0),
++
++SOC_DOUBLE_R("Headphone Playback Volume", WM8750_LOUT1V, WM8750_ROUT1V,
++ 0, 127, 0),
++SOC_DOUBLE_R("Speaker Playback Volume", WM8750_LOUT2V, WM8750_ROUT2V,
++ 0, 127, 0),
++
++SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
++
++};
++
++/* add non dapm controls */
++static int wm8750_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++ return 0;
++}
++
++/*
++ * DAPM Controls
++ */
++
++/* Left Mixer */
++static const struct snd_kcontrol_new wm8750_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Playback Switch", WM8750_LOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_LOUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8750_LOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_LOUTM2, 7, 1, 0),
++};
++
++/* Right Mixer */
++static const struct snd_kcontrol_new wm8750_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8750_ROUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_ROUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Playback Switch", WM8750_ROUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_ROUTM2, 7, 1, 0),
++};
++
++/* Mono Mixer */
++static const struct snd_kcontrol_new wm8750_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8750_MOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_MOUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8750_MOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_MOUTM2, 7, 1, 0),
++};
++
++/* Left Line Mux */
++static const struct snd_kcontrol_new wm8750_left_line_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[8]);
++
++/* Right Line Mux */
++static const struct snd_kcontrol_new wm8750_right_line_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[9]);
++
++/* Left PGA Mux */
++static const struct snd_kcontrol_new wm8750_left_pga_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[10]);
++
++/* Right PGA Mux */
++static const struct snd_kcontrol_new wm8750_right_pga_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[11]);
++
++/* Out 3 Mux */
++static const struct snd_kcontrol_new wm8750_out3_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[12]);
++
++/* Differential Mux */
++static const struct snd_kcontrol_new wm8750_diffmux_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[13]);
++
++/* Mono ADC Mux */
++static const struct snd_kcontrol_new wm8750_monomux_controls =
++SOC_DAPM_ENUM("Route", wm8750_enum[16]);
++
++static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
++ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8750_left_mixer_controls[0],
++ ARRAY_SIZE(wm8750_left_mixer_controls)),
++ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8750_right_mixer_controls[0],
++ ARRAY_SIZE(wm8750_right_mixer_controls)),
++ SND_SOC_DAPM_MIXER("Mono Mixer", WM8750_PWR2, 2, 0,
++ &wm8750_mono_mixer_controls[0],
++ ARRAY_SIZE(wm8750_mono_mixer_controls)),
++
++ SND_SOC_DAPM_PGA("Right Out 2", WM8750_PWR2, 3, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Left Out 2", WM8750_PWR2, 4, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Right Out 1", WM8750_PWR2, 5, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Left Out 1", WM8750_PWR2, 6, 0, NULL, 0),
++ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8750_PWR2, 7, 0),
++ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8750_PWR2, 8, 0),
++
++ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8750_PWR1, 1, 0),
++ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8750_PWR1, 2, 0),
++ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8750_PWR1, 3, 0),
++
++ SND_SOC_DAPM_MUX("Left PGA Mux", WM8750_PWR1, 5, 0,
++ &wm8750_left_pga_controls),
++ SND_SOC_DAPM_MUX("Right PGA Mux", WM8750_PWR1, 4, 0,
++ &wm8750_right_pga_controls),
++ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
++ &wm8750_left_line_controls),
++ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
++ &wm8750_right_line_controls),
++
++ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8750_out3_controls),
++ SND_SOC_DAPM_PGA("Out 3", WM8750_PWR2, 1, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Mono Out 1", WM8750_PWR2, 2, 0, NULL, 0),
++
++ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
++ &wm8750_diffmux_controls),
++ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
++ &wm8750_monomux_controls),
++ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
++ &wm8750_monomux_controls),
++
++ SND_SOC_DAPM_OUTPUT("LOUT1"),
++ SND_SOC_DAPM_OUTPUT("ROUT1"),
++ SND_SOC_DAPM_OUTPUT("LOUT2"),
++ SND_SOC_DAPM_OUTPUT("ROUT2"),
++ SND_SOC_DAPM_OUTPUT("MONO"),
++ SND_SOC_DAPM_OUTPUT("OUT3"),
++
++ SND_SOC_DAPM_INPUT("LINPUT1"),
++ SND_SOC_DAPM_INPUT("LINPUT2"),
++ SND_SOC_DAPM_INPUT("LINPUT3"),
++ SND_SOC_DAPM_INPUT("RINPUT1"),
++ SND_SOC_DAPM_INPUT("RINPUT2"),
++ SND_SOC_DAPM_INPUT("RINPUT3"),
++};
++
++static const char *audio_map[][3] = {
++ /* left mixer */
++ {"Left Mixer", "Playback Switch", "Left DAC"},
++ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
++ {"Left Mixer", "Right Playback Switch", "Right DAC"},
++ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
++
++ /* right mixer */
++ {"Right Mixer", "Left Playback Switch", "Left DAC"},
++ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
++ {"Right Mixer", "Playback Switch", "Right DAC"},
++ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
++
++ /* left out 1 */
++ {"Left Out 1", NULL, "Left Mixer"},
++ {"LOUT1", NULL, "Left Out 1"},
++
++ /* left out 2 */
++ {"Left Out 2", NULL, "Left Mixer"},
++ {"LOUT2", NULL, "Left Out 2"},
++
++ /* right out 1 */
++ {"Right Out 1", NULL, "Right Mixer"},
++ {"ROUT1", NULL, "Right Out 1"},
++
++ /* right out 2 */
++ {"Right Out 2", NULL, "Right Mixer"},
++ {"ROUT2", NULL, "Right Out 2"},
++
++ /* mono mixer */
++ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
++ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
++ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
++ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
++
++ /* mono out */
++ {"Mono Out 1", NULL, "Mono Mixer"},
++ {"MONO1", NULL, "Mono Out 1"},
++
++ /* out 3 */
++ {"Out3 Mux", "VREF", "VREF"},
++ {"Out3 Mux", "ROUT1 + Vol", "ROUT1"},
++ {"Out3 Mux", "ROUT1", "Right Mixer"},
++ {"Out3 Mux", "MonoOut", "MONO1"},
++ {"Out 3", NULL, "Out3 Mux"},
++ {"OUT3", NULL, "Out 3"},
++
++ /* Left Line Mux */
++ {"Left Line Mux", "Line 1", "LINPUT1"},
++ {"Left Line Mux", "Line 2", "LINPUT2"},
++ {"Left Line Mux", "Line 3", "LINPUT3"},
++ {"Left Line Mux", "PGA", "Left PGA Mux"},
++ {"Left Line Mux", "Differential", "Differential Mux"},
++
++ /* Right Line Mux */
++ {"Right Line Mux", "Line 1", "RINPUT1"},
++ {"Right Line Mux", "Line 2", "RINPUT2"},
++ {"Right Line Mux", "Line 3", "RINPUT3"},
++ {"Right Line Mux", "PGA", "Right PGA Mux"},
++ {"Right Line Mux", "Differential", "Differential Mux"},
++
++ /* Left PGA Mux */
++ {"Left PGA Mux", "Line 1", "LINPUT1"},
++ {"Left PGA Mux", "Line 2", "LINPUT2"},
++ {"Left PGA Mux", "Line 3", "LINPUT3"},
++ {"Left PGA Mux", "Differential", "Differential Mux"},
++
++ /* Right PGA Mux */
++ {"Right PGA Mux", "Line 1", "RINPUT1"},
++ {"Right PGA Mux", "Line 2", "RINPUT2"},
++ {"Right PGA Mux", "Line 3", "RINPUT3"},
++ {"Right PGA Mux", "Differential", "Differential Mux"},
++
++ /* Differential Mux */
++ {"Differential Mux", "Line 1", "LINPUT1"},
++ {"Differential Mux", "Line 1", "RINPUT1"},
++ {"Differential Mux", "Line 2", "LINPUT2"},
++ {"Differential Mux", "Line 2", "RINPUT2"},
++
++ /* Left ADC Mux */
++ {"Left ADC Mux", "Stereo", "Left PGA Mux"},
++ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
++ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
++
++ /* Right ADC Mux */
++ {"Right ADC Mux", "Stereo", "Right PGA Mux"},
++ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
++ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
++
++ /* ADC */
++ {"Left ADC", NULL, "Left ADC Mux"},
++ {"Right ADC", NULL, "Right ADC Mux"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int wm8750_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++struct _coeff_div {
++ u32 mclk;
++ u32 rate;
++ u16 fs;
++ u8 sr:5;
++ u8 usb:1;
++};
++
++/* codec hifi mclk clock divider coefficients */
++static const struct _coeff_div coeff_div[] = {
++ /* 8k */
++ {12288000, 8000, 1536, 0x6, 0x0},
++ {11289600, 8000, 1408, 0x16, 0x0},
++ {18432000, 8000, 2304, 0x7, 0x0},
++ {16934400, 8000, 2112, 0x17, 0x0},
++ {12000000, 8000, 1500, 0x6, 0x1},
++
++ /* 11.025k */
++ {11289600, 11025, 1024, 0x18, 0x0},
++ {16934400, 11025, 1536, 0x19, 0x0},
++ {12000000, 11025, 1088, 0x19, 0x1},
++
++ /* 16k */
++ {12288000, 16000, 768, 0xa, 0x0},
++ {18432000, 16000, 1152, 0xb, 0x0},
++ {12000000, 16000, 750, 0xa, 0x1},
++
++ /* 22.05k */
++ {11289600, 22050, 512, 0x1a, 0x0},
++ {16934400, 22050, 768, 0x1b, 0x0},
++ {12000000, 22050, 544, 0x1b, 0x1},
++
++ /* 32k */
++ {12288000, 32000, 384, 0xc, 0x0},
++ {18432000, 32000, 576, 0xd, 0x0},
++ {12000000, 32000, 375, 0xa, 0x1},
++
++ /* 44.1k */
++ {11289600, 44100, 256, 0x10, 0x0},
++ {16934400, 44100, 384, 0x11, 0x0},
++ {12000000, 44100, 272, 0x11, 0x1},
++
++ /* 48k */
++ {12288000, 48000, 256, 0x0, 0x0},
++ {18432000, 48000, 384, 0x1, 0x0},
++ {12000000, 48000, 250, 0x0, 0x1},
++
++ /* 88.2k */
++ {11289600, 88200, 128, 0x1e, 0x0},
++ {16934400, 88200, 192, 0x1f, 0x0},
++ {12000000, 88200, 136, 0x1f, 0x1},
++
++ /* 96k */
++ {12288000, 96000, 128, 0xe, 0x0},
++ {18432000, 96000, 192, 0xf, 0x0},
++ {12000000, 96000, 125, 0xe, 0x1},
++};
++
++static inline int get_coeff(int mclk, int rate)
++{
++ int i;
++
++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
++ return i;
++ }
++
++ printk(KERN_ERR "wm8750: could not get coeff for mclk %d @ rate %d\n",
++ mclk, rate);
++ return -EINVAL;
++}
++
++/* WM8750 supports numerous input clocks per sample rate */
++static unsigned int wm8750_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ dai->mclk = clk;
++ return dai->mclk;
++}
++
++static int wm8750_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 iface = 0, bfs, srate = 0;
++ int i = get_coeff(rtd->codec_dai->mclk,
++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate));
++
++ /* is coefficient valid ? */
++ if (i < 0)
++ return i;
++
++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
++
++ /* set master/slave audio interface */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ iface = 0x0040;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ break;
++ }
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ iface |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ iface |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ iface |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ iface |= 0x0013;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ iface |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ iface |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ iface |= 0x000c;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ break;
++ case SND_SOC_DAIFMT_IB_IF:
++ iface |= 0x0090;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ iface |= 0x0080;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ iface |= 0x0010;
++ break;
++ }
++
++ /* set bclk divisor rate */
++ switch (bfs) {
++ case 1:
++ break;
++ case 4:
++ srate |= (0x1 << 7);
++ break;
++ case 8:
++ srate |= (0x2 << 7);
++ break;
++ case 16:
++ srate |= (0x3 << 7);
++ break;
++ }
++
++ /* set iface & srate */
++ wm8750_write(codec, WM8750_IFACE, iface);
++ wm8750_write(codec, WM8750_SRATE, srate |
++ (coeff_div[i].sr << 1) | coeff_div[i].usb);
++
++ return 0;
++}
++
++static int wm8750_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
++ if (mute)
++ wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8);
++ else
++ wm8750_write(codec, WM8750_ADCDAC, mute_reg);
++ return 0;
++}
++
++static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* set vmid to 50k and unmute dac */
++ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ /* set vmid to 5k for quick power up */
++ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* mute dac and set vmid to 500k, enable VREF */
++ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ wm8750_write(codec, WM8750_PWR1, 0x0001);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai wm8750_dai = {
++ .name = "WM8750",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = wm8750_config_sysclk,
++ .digital_mute = wm8750_mute,
++ .ops = {
++ .prepare = wm8750_pcm_prepare,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8750_modes),
++ .mode = wm8750_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(wm8750_dai);
++
++static void wm8750_work(void *data)
++{
++ struct snd_soc_codec *codec = (struct snd_soc_codec *)data;
++ wm8750_dapm_event(codec, codec->dapm_state);
++}
++
++static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm8750_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) {
++ if (i == WM8750_RESET)
++ continue;
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++
++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* charge wm8750 caps */
++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
++ codec->dapm_state = SNDRV_CTL_POWER_D0;
++ queue_delayed_work(wm8750_workq, &wm8750_dapm_work,
++ msecs_to_jiffies(1000));
++ }
++
++ return 0;
++}
++
++/*
++ * initialise the WM8750 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8750_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg, ret = 0;
++
++ codec->name = "WM8750";
++ codec->owner = THIS_MODULE;
++ codec->read = wm8750_read_reg_cache;
++ codec->write = wm8750_write;
++ codec->dapm_event = wm8750_dapm_event;
++ codec->dai = &wm8750_dai;
++ codec->num_dai = 1;
++ codec->reg_cache_size = ARRAY_SIZE(wm8750_reg);
++
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8750_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, wm8750_reg,
++ sizeof(u16) * ARRAY_SIZE(wm8750_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8750_reg);
++
++ wm8750_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* charge output caps */
++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
++ codec->dapm_state = SNDRV_CTL_POWER_D3hot;
++ queue_delayed_work(wm8750_workq, &wm8750_dapm_work,
++ msecs_to_jiffies(1000));
++
++ /* set the update bits */
++ reg = wm8750_read_reg_cache(codec, WM8750_LDAC);
++ wm8750_write(codec, WM8750_LDAC, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_RDAC);
++ wm8750_write(codec, WM8750_RDAC, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V);
++ wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V);
++ wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V);
++ wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V);
++ wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_LINVOL);
++ wm8750_write(codec, WM8750_LINVOL, reg | 0x0100);
++ reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
++ wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
++
++ wm8750_add_controls(codec);
++ wm8750_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++static struct snd_soc_device *wm8750_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8731 2 wire address is determined by GPIO5
++ * state during powerup.
++ * low = 0x1a
++ * high = 0x1b
++ */
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver wm8750_i2c_driver;
++static struct i2c_client client_template;
++
++static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = wm8750_socdev;
++ struct wm8750_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if (ret < 0) {
++ err("failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = wm8750_init(socdev);
++ if (ret < 0) {
++ err("failed to initialise WM8750\n");
++ goto err;
++ }
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int wm8750_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec *codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int wm8750_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, wm8750_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8750_i2c_driver = {
++ .driver = {
++ .name = "WM8750 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_WM8750,
++ .attach_adapter = wm8750_i2c_attach,
++ .detach_client = wm8750_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "WM8750",
++ .driver = &wm8750_i2c_driver,
++};
++#endif
++
++static int wm8750_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct wm8750_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ info("WM8750 Audio Codec %s", WM8750_VERSION);
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++ wm8750_socdev = socdev;
++ INIT_WORK(&wm8750_dapm_work, wm8750_work, codec);
++ wm8750_workq = create_workqueue("wm8750");
++ if (wm8750_workq == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&wm8750_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++
++ return ret;
++}
++
++/* power down chip */
++static int wm8750_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec->control_data)
++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ if (wm8750_workq)
++ destroy_workqueue(wm8750_workq);
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&wm8750_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8750 = {
++ .probe = wm8750_probe,
++ .remove = wm8750_remove,
++ .suspend = wm8750_suspend,
++ .resume = wm8750_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
++
++MODULE_DESCRIPTION("ASoC WM8750 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8750.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8750.h
+@@ -0,0 +1,66 @@
++/*
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard@openedhand.com>
++ *
++ * Based on WM8753.h
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ */
++
++#ifndef _WM8750_H
++#define _WM8750_H
++
++/* WM8750 register space */
++
++#define WM8750_LINVOL 0x00
++#define WM8750_RINVOL 0x01
++#define WM8750_LOUT1V 0x02
++#define WM8750_ROUT1V 0x03
++#define WM8750_ADCDAC 0x05
++#define WM8750_IFACE 0x07
++#define WM8750_SRATE 0x08
++#define WM8750_LDAC 0x0a
++#define WM8750_RDAC 0x0b
++#define WM8750_BASS 0x0c
++#define WM8750_TREBLE 0x0d
++#define WM8750_RESET 0x0f
++#define WM8750_3D 0x10
++#define WM8750_ALC1 0x11
++#define WM8750_ALC2 0x12
++#define WM8750_ALC3 0x13
++#define WM8750_NGATE 0x14
++#define WM8750_LADC 0x15
++#define WM8750_RADC 0x16
++#define WM8750_ADCTL1 0x17
++#define WM8750_ADCTL2 0x18
++#define WM8750_PWR1 0x19
++#define WM8750_PWR2 0x1a
++#define WM8750_ADCTL3 0x1b
++#define WM8750_ADCIN 0x1f
++#define WM8750_LADCIN 0x20
++#define WM8750_RADCIN 0x21
++#define WM8750_LOUTM1 0x22
++#define WM8750_LOUTM2 0x23
++#define WM8750_ROUTM1 0x24
++#define WM8750_ROUTM2 0x25
++#define WM8750_MOUTM1 0x26
++#define WM8750_MOUTM2 0x27
++#define WM8750_LOUT2V 0x28
++#define WM8750_ROUT2V 0x29
++#define WM8750_MOUTV 0x2a
++
++#define WM8750_CACHE_REGNUM 0x2a
++
++struct wm8750_setup_data {
++ unsigned short i2c_address;
++ unsigned int mclk;
++};
++
++extern struct snd_soc_codec_dai wm8750_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8750;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8753.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8753.c
+@@ -0,0 +1,2128 @@
++/*
++ * wm8753.c -- WM8753 ALSA Soc Audio driver
++ *
++ * Copyright 2003 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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.
++ *
++ * Notes:
++ * The WM8753 is a low power, high quality stereo codec with integrated PCM
++ * codec designed for portable digital telephony applications.
++ *
++ * Dual DAI:-
++ *
++ * This driver support 2 DAI PCM's. This makes the default PCM available for
++ * HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for
++ * voice.
++ *
++ * Please note that the voice PCM can be connected directly to a Bluetooth
++ * codec or GSM modem and thus cannot be read or written to, although it is
++ * available to be configured with snd_hw_params(), etc and kcontrols in the
++ * normal alsa manner.
++ *
++ * Fast DAI switching:-
++ *
++ * The driver can now fast switch between the DAI configurations via a
++ * an alsa kcontrol. This allows the PCM to remain open.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8753.h"
++
++#define AUDIO_NAME "wm8753"
++#define WM8753_VERSION "0.16"
++
++/*
++ * Debug
++ */
++
++#define WM8753_DEBUG 0
++
++#ifdef WM8753_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++static int caps_charge = 2000;
++module_param(caps_charge, int, 0);
++MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
++
++static struct workqueue_struct *wm8753_workq = NULL;
++static struct work_struct wm8753_dapm_work;
++static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
++ unsigned int mode);
++
++/*
++ * wm8753 register cache
++ * We can't read the WM8753 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8753_reg[] = {
++ 0x0008, 0x0000, 0x000a, 0x000a,
++ 0x0033, 0x0000, 0x0007, 0x00ff,
++ 0x00ff, 0x000f, 0x000f, 0x007b,
++ 0x0000, 0x0032, 0x0000, 0x00c3,
++ 0x00c3, 0x00c0, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0055,
++ 0x0005, 0x0050, 0x0055, 0x0050,
++ 0x0055, 0x0050, 0x0055, 0x0079,
++ 0x0079, 0x0079, 0x0079, 0x0079,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0097, 0x0097, 0x0000, 0x0004,
++ 0x0000, 0x0083, 0x0024, 0x01ba,
++ 0x0000, 0x0083, 0x0024, 0x01ba,
++ 0x0000, 0x0000
++};
++
++#define WM8753_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \
++ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)
++
++#define WM8753_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM8753_HIFI_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++#define WM8753_HIFI_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8753_HIFI_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++/*
++ * HiFi modes
++ */
++static struct snd_soc_dai_mode wm8753_hifi_modes[] = {
++ /* codec frame and clock master modes */
++ /* 8k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1408,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 2304,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 2112,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1500,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* 11.025k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1024,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1088,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* 16k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 768,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt= WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1152,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 750,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* 22.05k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 512,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 768,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 544,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* 32k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 576,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 375,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* 44.1k & 48k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 250,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 272,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* 88.2k & 96k */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 128,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 192,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 136,
++ .bfs = WM8753_HIFI_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 125,
++ .bfs = WM8753_HIFI_FSB,
++ },
++
++ /* codec frame and clock slave modes */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = WM8753_HIFI_RATES,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++#define WM8753_VOICE_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++#define WM8753_VOICE_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++#define WM8753_VOICE_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++/*
++ * Voice modes
++ */
++static struct snd_soc_dai_mode wm8753_voice_modes[] = {
++
++ /* master modes */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_VOICE_BITS,
++ .pcmrate = WM8753_VOICE_RATES,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = WM8753_VOICE_FSB,
++ },
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM8753_VOICE_BITS,
++ .pcmrate = WM8753_VOICE_RATES,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8753_VOICE_FSB,
++ },
++
++ /* slave modes */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8753_VOICE_BITS,
++ .pcmrate = WM8753_VOICE_RATES,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++
++/*
++ * Mode 4
++ */
++static struct snd_soc_dai_mode wm8753_mixed_modes[] = {
++ /* slave modes */
++ {
++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8753_HIFI_BITS,
++ .pcmrate = WM8753_HIFI_RATES,
++ .pcmdir = WM8753_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++/*
++ * read wm8753 register cache
++ */
++static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
++ return -1;
++ return cache[reg - 1];
++}
++
++/*
++ * write wm8753 register cache
++ */
++static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg < 1 || reg > 0x3f)
++ return;
++ cache[reg - 1] = value;
++}
++
++/*
++ * write to the WM8753 register space
++ */
++static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D9 WM8753 register offset
++ * D8...D0 register data
++ */
++ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++ data[1] = value & 0x00ff;
++
++ wm8753_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -EIO;
++}
++
++#define wm8753_reset(c) wm8753_write(c, WM8753_RESET, 0)
++
++/*
++ * WM8753 Controls
++ */
++static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"};
++static const char *wm8753_base_filter[] =
++ {"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz",
++ "100Hz @ 8kHz", "200Hz @ 8kHz"};
++static const char *wm8753_treble[] = {"8kHz", "4kHz"};
++static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"};
++static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"};
++static const char *wm8753_3d_func[] = {"Capture", "Playback"};
++static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"};
++static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"};
++static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"};
++static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"};
++static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"};
++static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2",
++ "Line 1", "Line 2"};
++static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"};
++static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"};
++static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"};
++static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"};
++static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2",
++ "Right PGA"};
++static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right",
++ "Left + Right"};
++static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"};
++static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"};
++static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"};
++static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"};
++static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left",
++ "Analogue Mix Right", "Digital Mono Mix"};
++static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k",
++ "82Hz @ 8kHz", "170Hz @ 8kHz"};
++static const char *wm8753_adc_filter[] = {"HiFi", "Voice"};
++static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"};
++static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"};
++
++static const struct soc_enum wm8753_enum[] = {
++SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), // 0
++SOC_ENUM_SINGLE(WM8753_BASS, 4, 6, wm8753_base_filter), // 1
++SOC_ENUM_SINGLE(WM8753_TREBLE, 6, 2, wm8753_treble), // 2
++SOC_ENUM_SINGLE(WM8753_ALC1, 7, 4, wm8753_alc_func), // 3
++SOC_ENUM_SINGLE(WM8753_NGATE, 1, 2, wm8753_ng_type), // 4
++SOC_ENUM_SINGLE(WM8753_3D, 7, 2, wm8753_3d_func), // 5
++SOC_ENUM_SINGLE(WM8753_3D, 6, 2, wm8753_3d_uc), // 6
++SOC_ENUM_SINGLE(WM8753_3D, 5, 2, wm8753_3d_lc), // 7
++SOC_ENUM_SINGLE(WM8753_DAC, 1, 4, wm8753_deemp), // 8
++SOC_ENUM_SINGLE(WM8753_DAC, 4, 4, wm8753_mono_mix), // 9
++SOC_ENUM_SINGLE(WM8753_DAC, 6, 2, wm8753_dac_phase), // 10
++SOC_ENUM_SINGLE(WM8753_INCTL1, 3, 4, wm8753_line_mix), // 11
++SOC_ENUM_SINGLE(WM8753_INCTL1, 2, 2, wm8753_mono_mux), // 12
++SOC_ENUM_SINGLE(WM8753_INCTL1, 1, 2, wm8753_right_mux), // 13
++SOC_ENUM_SINGLE(WM8753_INCTL1, 0, 2, wm8753_left_mux), // 14
++SOC_ENUM_SINGLE(WM8753_INCTL2, 6, 4, wm8753_rxmsel), // 15
++SOC_ENUM_SINGLE(WM8753_INCTL2, 4, 4, wm8753_sidetone_mux),// 16
++SOC_ENUM_SINGLE(WM8753_OUTCTL, 7, 4, wm8753_mono2_src), // 17
++SOC_ENUM_SINGLE(WM8753_OUTCTL, 0, 3, wm8753_out3), // 18
++SOC_ENUM_SINGLE(WM8753_ADCTL2, 7, 3, wm8753_out4), // 19
++SOC_ENUM_SINGLE(WM8753_ADCIN, 2, 3, wm8753_radcsel), // 20
++SOC_ENUM_SINGLE(WM8753_ADCIN, 0, 3, wm8753_ladcsel), // 21
++SOC_ENUM_SINGLE(WM8753_ADCIN, 4, 4, wm8753_mono_adc), // 22
++SOC_ENUM_SINGLE(WM8753_ADC, 2, 4, wm8753_adc_hp), // 23
++SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), // 24
++SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), // 25
++SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), // 26
++};
++
++
++static int wm8753_get_dai(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
++
++ ucontrol->value.integer.value[0] = (mode & 0xc) >> 2;
++ return 0;
++}
++
++static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
++
++ if (((mode &0xc) >> 2) == ucontrol->value.integer.value[0])
++ return 0;
++
++ mode &= 0xfff3;
++ mode |= (ucontrol->value.integer.value[0] << 2);
++
++ wm8753_write(codec, WM8753_IOCTL, mode);
++ wm8753_set_dai_mode(codec, ucontrol->value.integer.value[0]);
++ return 1;
++}
++
++static const struct snd_kcontrol_new wm8753_snd_controls[] = {
++SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0),
++
++SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 63, 0),
++SOC_DOUBLE_R("ADC Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 0),
++SOC_DOUBLE_R("ADC Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
++
++SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0),
++SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0),
++
++SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0),
++
++SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1),
++SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1),
++SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1),
++
++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0),
++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0),
++
++SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
++SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
++SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1),
++SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
++
++SOC_ENUM("Bass Boost", wm8753_enum[0]),
++SOC_ENUM("Bass Filter", wm8753_enum[1]),
++SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 7, 1),
++
++SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 7, 0),
++SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
++
++SOC_DOUBLE("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1),
++SOC_SINGLE("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1),
++
++SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0),
++SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
++SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 0),
++
++SOC_ENUM("Capture Filter Select", wm8753_enum[23]),
++SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]),
++SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1),
++
++SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0),
++SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0),
++SOC_ENUM("ALC Capture Function", wm8753_enum[3]),
++SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0),
++SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1),
++SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1),
++SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0),
++SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0),
++SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]),
++SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0),
++
++SOC_ENUM("3D Function", wm8753_enum[5]),
++SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]),
++SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]),
++SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0),
++SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0),
++
++SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0),
++SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0),
++
++SOC_ENUM("De-emphasis", wm8753_enum[8]),
++SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
++SOC_ENUM("Playback Phase", wm8753_enum[10]),
++
++SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0),
++SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
++
++SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
++};
++
++/* add non dapm controls */
++static int wm8753_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++ return 0;
++}
++
++/*
++ * _DAPM_ Controls
++ */
++
++/* Left Mixer */
++static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0),
++SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0),
++};
++
++/* Right mixer */
++static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0),
++};
++
++/* Mono mixer */
++static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0),
++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0),
++};
++
++/* Mono 2 Mux */
++static const struct snd_kcontrol_new wm8753_mono2_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[17]);
++
++/* Out 3 Mux */
++static const struct snd_kcontrol_new wm8753_out3_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[18]);
++
++/* Out 4 Mux */
++static const struct snd_kcontrol_new wm8753_out4_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[19]);
++
++/* ADC Mono Mix */
++static const struct snd_kcontrol_new wm8753_adc_mono_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[22]);
++
++/* Record mixer */
++static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = {
++SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0),
++SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0),
++SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0),
++};
++
++/* Left ADC mux */
++static const struct snd_kcontrol_new wm8753_adc_left_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[21]);
++
++/* Right ADC mux */
++static const struct snd_kcontrol_new wm8753_adc_right_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[20]);
++
++/* MIC mux */
++static const struct snd_kcontrol_new wm8753_mic_mux_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[16]);
++
++/* ALC mixer */
++static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0),
++SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0),
++SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0),
++SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0),
++};
++
++/* Left Line mux */
++static const struct snd_kcontrol_new wm8753_line_left_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[14]);
++
++/* Right Line mux */
++static const struct snd_kcontrol_new wm8753_line_right_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[13]);
++
++/* Mono Line mux */
++static const struct snd_kcontrol_new wm8753_line_mono_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[12]);
++
++/* Line mux and mixer */
++static const struct snd_kcontrol_new wm8753_line_mux_mix_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[11]);
++
++/* Rx mux and mixer */
++static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[15]);
++
++/* Mic Selector Mux */
++static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[25]);
++
++static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0),
++SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0,
++ &wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)),
++SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0),
++SND_SOC_DAPM_OUTPUT("LOUT1"),
++SND_SOC_DAPM_OUTPUT("LOUT2"),
++SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0,
++ &wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)),
++SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0),
++SND_SOC_DAPM_OUTPUT("ROUT1"),
++SND_SOC_DAPM_OUTPUT("ROUT2"),
++SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0,
++ &wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)),
++SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0),
++SND_SOC_DAPM_OUTPUT("MONO1"),
++SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls),
++SND_SOC_DAPM_OUTPUT("MONO2"),
++SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0),
++SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls),
++SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0),
++SND_SOC_DAPM_OUTPUT("OUT3"),
++SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls),
++SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0),
++SND_SOC_DAPM_OUTPUT("OUT4"),
++SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0,
++ &wm8753_record_mixer_controls[0],
++ ARRAY_SIZE(wm8753_record_mixer_controls)),
++SND_SOC_DAPM_ADC("Left ADC", "Left Voice Capture", WM8753_PWR2, 3, 0),
++SND_SOC_DAPM_ADC("Right ADC", "Right Voice Capture", WM8753_PWR2, 2, 0),
++SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8753_adc_mono_controls),
++SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8753_adc_mono_controls),
++SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_adc_left_controls),
++SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_adc_right_controls),
++SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_mic_mux_controls),
++SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0),
++SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0,
++ &wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)),
++SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_line_left_controls),
++SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_line_right_controls),
++SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_line_mono_controls),
++SND_SOC_DAPM_MUX("Line Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8753_line_mux_mix_controls),
++SND_SOC_DAPM_MUX("Rx Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8753_rx_mux_mix_controls),
++SND_SOC_DAPM_PGA("Mic 1 Volume", WM8753_PWR2, 8, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mic 2 Volume", WM8753_PWR2, 7, 0, NULL, 0),
++SND_SOC_DAPM_MUX("Mic Selection Mux", SND_SOC_NOPM, 0, 0,
++ &wm8753_mic_sel_mux_controls),
++SND_SOC_DAPM_INPUT("LINE1"),
++SND_SOC_DAPM_INPUT("LINE2"),
++SND_SOC_DAPM_INPUT("RXP"),
++SND_SOC_DAPM_INPUT("RXN"),
++SND_SOC_DAPM_INPUT("ACIN"),
++SND_SOC_DAPM_INPUT("ACOP"),
++SND_SOC_DAPM_INPUT("MIC1N"),
++SND_SOC_DAPM_INPUT("MIC1"),
++SND_SOC_DAPM_INPUT("MIC2N"),
++SND_SOC_DAPM_INPUT("MIC2"),
++SND_SOC_DAPM_VMID("VREF"),
++};
++
++static const char *audio_map[][3] = {
++ /* left mixer */
++ {"Left Mixer", "Left Playback Switch", "Left DAC"},
++ {"Left Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
++ {"Left Mixer", "Bypass Playback Switch", "Line Left Mux"},
++
++ /* right mixer */
++ {"Right Mixer", "Right Playback Switch", "Right DAC"},
++ {"Right Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
++ {"Right Mixer", "Bypass Playback Switch", "Line Right Mux"},
++
++ /* mono mixer */
++ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
++ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
++ {"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
++ {"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"},
++
++ /* left out */
++ {"Left Out 1", NULL, "Left Mixer"},
++ {"Left Out 2", NULL, "Left Mixer"},
++ {"LOUT1", NULL, "Left Out 1"},
++ {"LOUT2", NULL, "Left Out 2"},
++
++ /* right out */
++ {"Right Out 1", NULL, "Right Mixer"},
++ {"Right Out 2", NULL, "Right Mixer"},
++ {"ROUT1", NULL, "Right Out 1"},
++ {"ROUT2", NULL, "Right Out 2"},
++
++ /* mono 1 out */
++ {"Mono Out 1", NULL, "Mono Mixer"},
++ {"MONO1", NULL, "Mono Out 1"},
++
++ /* mono 2 out */
++ {"Mono 2 Mux", "Left + Right", "Out3 Left + Right"},
++ {"Mono 2 Mux", "Inverted Mono 1", "MONO1"},
++ {"Mono 2 Mux", "Left", "Left Mixer"},
++ {"Mono 2 Mux", "Right", "Right Mixer"},
++ {"Mono Out 2", NULL, "Mono 2 Mux"},
++ {"MONO2", NULL, "Mono Out 2"},
++
++ /* out 3 */
++ {"Out3 Left + Right", NULL, "Left Mixer"},
++ {"Out3 Left + Right", NULL, "Right Mixer"},
++ {"Out3 Mux", "VREF", "VREF"},
++ {"Out3 Mux", "Left + Right", "Out3 Left + Right"},
++ {"Out3 Mux", "ROUT2", "ROUT2"},
++ {"Out 3", NULL, "Out3 Mux"},
++ {"OUT3", NULL, "Out 3"},
++
++ /* out 4 */
++ {"Out4 Mux", "VREF", "VREF"},
++ {"Out4 Mux", "Capture ST", "Capture ST Mixer"},
++ {"Out4 Mux", "LOUT2", "LOUT2"},
++ {"Out 4", NULL, "Out4 Mux"},
++ {"OUT4", NULL, "Out 4"},
++
++ /* record mixer */
++ {"Playback Mixer", "Left Capture Switch", "Left Mixer"},
++ {"Playback Mixer", "Voice Capture Switch", "Mono Mixer"},
++ {"Playback Mixer", "Right Capture Switch", "Right Mixer"},
++
++ /* Mic/SideTone Mux */
++ {"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"},
++ {"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"},
++ {"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"},
++ {"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"},
++
++ /* Capture Left Mux */
++ {"Capture Left Mux", "PGA", "Left Capture Volume"},
++ {"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"},
++ {"Capture Left Mux", "Line", "LINE1"},
++
++ /* Capture Right Mux */
++ {"Capture Right Mux", "PGA", "Right Capture Volume"},
++ {"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
++ {"Capture Right Mux", "Sidetone", "Capture ST Mixer"},
++
++ /* Mono Capture mixer-mux */
++ {"Capture Right Mixer", "Stereo", "Capture Right Mux"},
++ {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
++ {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
++ {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
++ {"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"},
++ {"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"},
++ {"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"},
++ {"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"},
++ {"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"},
++
++ /* ADC */
++ {"Left ADC", NULL, "Capture Left Mixer"},
++ {"Right ADC", NULL, "Capture Right Mixer"},
++
++ /* Left Capture Volume */
++ {"Left Capture Volume", NULL, "ACIN"},
++
++ /* Right Capture Volume */
++ {"Right Capture Volume", NULL, "Mic 2 Volume"},
++
++ /* ALC Mixer */
++ {"ALC Mixer", "Line Capture Switch", "Line Mixer"},
++ {"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"},
++ {"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"},
++ {"ALC Mixer", "Rx Capture Switch", "Rx Mixer"},
++
++ /* Line Left Mux */
++ {"Line Left Mux", "Line 1", "LINE1"},
++ {"Line Left Mux", "Rx Mix", "Rx Mixer"},
++
++ /* Line Right Mux */
++ {"Line Right Mux", "Line 2", "LINE2"},
++ {"Line Right Mux", "Rx Mix", "Rx Mixer"},
++
++ /* Line Mono Mux */
++ {"Line Mono Mux", "Line Mix", "Line Mixer"},
++ {"Line Mono Mux", "Rx Mix", "Rx Mixer"},
++
++ /* Line Mixer/Mux */
++ {"Line Mixer", "Line 1 + 2", "LINE1"},
++ {"Line Mixer", "Line 1 - 2", "LINE1"},
++ {"Line Mixer", "Line 1 + 2", "LINE2"},
++ {"Line Mixer", "Line 1 - 2", "LINE2"},
++ {"Line Mixer", "Line 1", "LINE1"},
++ {"Line Mixer", "Line 2", "LINE2"},
++
++ /* Rx Mixer/Mux */
++ {"Rx Mixer", "RXP - RXN", "RXP"},
++ {"Rx Mixer", "RXP + RXN", "RXP"},
++ {"Rx Mixer", "RXP - RXN", "RXN"},
++ {"Rx Mixer", "RXP + RXN", "RXN"},
++ {"Rx Mixer", "RXP", "RXP"},
++ {"Rx Mixer", "RXN", "RXN"},
++
++ /* Mic 1 Volume */
++ {"Mic 1 Volume", NULL, "MIC1N"},
++ {"Mic 1 Volume", NULL, "Mic Selection Mux"},
++
++ /* Mic 2 Volume */
++ {"Mic 2 Volume", NULL, "MIC2N"},
++ {"Mic 2 Volume", NULL, "MIC2"},
++
++ /* Mic Selector Mux */
++ {"Mic Selection Mux", "Mic 1", "MIC1"},
++ {"Mic Selection Mux", "Mic 2", "MIC2N"},
++ {"Mic Selection Mux", "Mic 3", "MIC2"},
++
++ /* ACOP */
++ {"ACOP", NULL, "ALC Mixer"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int wm8753_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
++ }
++
++ /* set up the WM8753 audio map */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++/* PLL divisors */
++struct _pll_div {
++ u32 pll_in; /* ext clock input */
++ u32 pll_out; /* pll out freq */
++ u32 div2:1;
++ u32 n:4;
++ u32 k:24;
++};
++
++/*
++ * PLL divisors -
++ */
++static const struct _pll_div pll_div[] = {
++ {13000000, 12288000, 0, 0x7, 0x23F54A},
++ {13000000, 11289600, 0, 0x6, 0x3CA2F5},
++ {12000000, 12288000, 0, 0x8, 0x0C49BA},
++ {12000000, 11289600, 0, 0x7, 0x21B08A},
++ {24000000, 12288000, 1, 0x8, 0x0C49BA},
++ {24000000, 11289600, 1, 0x7, 0x21B08A},
++ {12288000, 11289600, 0, 0x7, 0x166667},
++ {26000000, 11289600, 1, 0x6, 0x3CA2F5},
++ {26000000, 12288000, 1, 0x7, 0x23F54A},
++};
++
++static u32 wm8753_config_pll(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int pll)
++{
++ u16 reg;
++ int found = 0;
++
++ if (pll == 1) {
++ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xffef;
++ if (!dai->pll_in || !dai->mclk) {
++ /* disable PLL1 */
++ wm8753_write(codec, WM8753_PLL1CTL1, 0x0026);
++ wm8753_write(codec, WM8753_CLOCK, reg);
++ return 0;
++ } else {
++ u16 value = 0;
++ int i = 0;
++
++ /* if we cant match, then use good values for N and K */
++ for (;i < ARRAY_SIZE(pll_div); i++) {
++ if (pll_div[i].pll_out == dai->pll_out &&
++ pll_div[i].pll_in == dai->pll_in) {
++ found = 1;
++ break;
++ }
++ }
++
++ if (!found)
++ goto err;
++
++ /* set up N and K PLL divisor ratios */
++ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
++ value = (pll_div[i].n << 5) + ((pll_div[i].k & 0x3c0000) >> 18);
++ wm8753_write(codec, WM8753_PLL1CTL2, value);
++
++ /* bits 8:0 = PLL_K[17:9] */
++ value = (pll_div[i].k & 0x03fe00) >> 9;
++ wm8753_write(codec, WM8753_PLL1CTL3, value);
++
++ /* bits 8:0 = PLL_K[8:0] */
++ value = pll_div[i].k & 0x0001ff;
++ wm8753_write(codec, WM8753_PLL1CTL4, value);
++
++ /* set PLL1 as input and enable */
++ wm8753_write(codec, WM8753_PLL1CTL1, 0x0027 |
++ (pll_div[i].div2 << 3));
++ wm8753_write(codec, WM8753_CLOCK, reg | 0x0010);
++ }
++ } else {
++ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfff7;
++ if (!dai->pll_in || !dai->mclk) {
++ /* disable PLL2 */
++ wm8753_write(codec, WM8753_PLL2CTL1, 0x0026);
++ wm8753_write(codec, WM8753_CLOCK, reg);
++ return 0;
++ } else {
++ u16 value = 0;
++ int i = 0;
++
++ /* if we cant match, then use good values for N and K */
++ for (;i < ARRAY_SIZE(pll_div); i++) {
++ if (pll_div[i].pll_out == dai->pll_out &&
++ pll_div[i].pll_in == dai->pll_in) {
++ found = 1;
++ break;
++ }
++ }
++
++ if (!found)
++ goto err;
++
++ /* set up N and K PLL divisor ratios */
++ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
++ value = (pll_div[i].n << 5) + ((pll_div[i].k & 0x3c0000) >> 18);
++ wm8753_write(codec, WM8753_PLL2CTL2, value);
++
++ /* bits 8:0 = PLL_K[17:9] */
++ value = (pll_div[i].k & 0x03fe00) >> 9;
++ wm8753_write(codec, WM8753_PLL2CTL3, value);
++
++ /* bits 8:0 = PLL_K[8:0] */
++ value = pll_div[i].k & 0x0001ff;
++ wm8753_write(codec, WM8753_PLL2CTL4, value);
++
++ /* set PLL1 as input and enable */
++ wm8753_write(codec, WM8753_PLL2CTL1, 0x0027 |
++ (pll_div[i].div2 << 3));
++ wm8753_write(codec, WM8753_CLOCK, reg | 0x0008);
++ }
++ }
++
++ return dai->pll_in;
++err:
++ return 0;
++}
++
++struct _coeff_div {
++ u32 mclk;
++ u32 rate;
++ u16 fs;
++ u8 sr:5;
++ u8 usb:1;
++};
++
++/* codec hifi mclk (after PLL) clock divider coefficients */
++static const struct _coeff_div coeff_div[] = {
++ /* 8k */
++ {12288000, 8000, 1536, 0x6, 0x0},
++ {11289600, 8000, 1408, 0x16, 0x0},
++ {18432000, 8000, 2304, 0x7, 0x0},
++ {16934400, 8000, 2112, 0x17, 0x0},
++ {12000000, 8000, 1500, 0x6, 0x1},
++
++ /* 11.025k */
++ {11289600, 11025, 1024, 0x18, 0x0},
++ {16934400, 11025, 1536, 0x19, 0x0},
++ {12000000, 11025, 1088, 0x19, 0x1},
++
++ /* 16k */
++ {12288000, 16000, 768, 0xa, 0x0},
++ {18432000, 16000, 1152, 0xb, 0x0},
++ {12000000, 16000, 750, 0xa, 0x1},
++
++ /* 22.05k */
++ {11289600, 22050, 512, 0x1a, 0x0},
++ {16934400, 22050, 768, 0x1b, 0x0},
++ {12000000, 22050, 544, 0x1b, 0x1},
++
++ /* 32k */
++ {12288000, 32000, 384, 0xc, 0x0},
++ {18432000, 32000, 576, 0xd, 0x0},
++ {12000000, 32000, 375, 0xa, 0x1},
++
++ /* 44.1k */
++ {11289600, 44100, 256, 0x10, 0x0},
++ {16934400, 44100, 384, 0x11, 0x0},
++ {12000000, 44100, 272, 0x11, 0x1},
++
++ /* 48k */
++ {12288000, 48000, 256, 0x0, 0x0},
++ {18432000, 48000, 384, 0x1, 0x0},
++ {12000000, 48000, 250, 0x0, 0x1},
++
++ /* 88.2k */
++ {11289600, 88200, 128, 0x1e, 0x0},
++ {16934400, 88200, 192, 0x1f, 0x0},
++ {12000000, 88200, 136, 0x1f, 0x1},
++
++ /* 96k */
++ {12288000, 96000, 128, 0xe, 0x0},
++ {18432000, 96000, 192, 0xf, 0x0},
++ {12000000, 96000, 125, 0xe, 0x1},
++};
++
++static int get_coeff(int mclk, int rate)
++{
++ int i;
++
++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
++ return i;
++ }
++ return -EINVAL;
++}
++
++/* supported HiFi input clocks (that don't use PLL) */
++const static int hifi_clks[] = {11289600, 12000000, 12288000,
++ 16934400, 18432000};
++
++/* The HiFi interface can be clocked in one of two ways:-
++ * o No PLL - MCLK is used directly.
++ * o PLL - PLL is used to generate audio MCLK from input clock.
++ *
++ * We use the direct method if we can as it saves power.
++ */
++static unsigned int wm8753_config_i2s_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ int i, pll_out;
++
++ /* is clk supported without the PLL */
++ for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) {
++ if (clk == hifi_clks[i]) {
++ dai->mclk = clk;
++ dai->pll_in = dai->pll_out = 0;
++ dai->clk_div = 1;
++ return clk;
++ }
++ }
++
++ /* determine best PLL output speed */
++ if (info->bclk_master &
++ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
++ pll_out = info->fs * info->rate;
++ } else {
++ /* calc slave clock */
++ switch (info->rate){
++ case 11025:
++ case 22050:
++ case 44100:
++ case 88200:
++ pll_out = 11289600;
++ break;
++ default:
++ pll_out = 12288000;
++ break;
++ }
++ }
++
++ /* are input & output clocks supported by PLL */
++ for (i = 0;i < ARRAY_SIZE(pll_div); i++) {
++ if (pll_div[i].pll_in == clk && pll_div[i].pll_out == pll_out) {
++ dai->pll_in = clk;
++ dai->pll_out = dai->mclk = pll_out;
++ return pll_out;
++ }
++ }
++
++ /* this clk is not supported */
++ return 0;
++}
++
++/* valid PCM clock dividers * 2 */
++static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16};
++
++/* The Voice interface can be clocked in one of four ways:-
++ * o No PLL - MCLK is used directly.
++ * o Div - MCLK is directly divided.
++ * o PLL - PLL is used to generate audio MCLK from input clock.
++ * o PLL & Div - PLL and post divider are used.
++ *
++ * We use the non PLL methods if we can, as it saves power.
++ */
++
++static unsigned int wm8753_config_pcm_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ int i, j, best_clk = info->fs * info->rate;
++
++ /* can we run at this clk without the PLL ? */
++ for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) {
++ if ((best_clk >> 1) * pcm_divs[i] == clk) {
++ dai->pll_in = 0;
++ dai->clk_div = pcm_divs[i];
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++
++ /* now check for PLL support */
++ for (i = 0; i < ARRAY_SIZE(pll_div); i++) {
++ if (pll_div[i].pll_in == clk) {
++ for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) {
++ if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) {
++ dai->pll_in = clk;
++ dai->pll_out = pll_div[i].pll_out;
++ dai->clk_div = pcm_divs[j];
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++ }
++ }
++
++ /* this clk is not supported */
++ return 0;
++}
++
++/* set the format and bit size for ADC and Voice DAC */
++static void wm8753_adc_vdac_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01e0;
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ voice |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ voice |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ voice |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ voice |= 0x0013;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ voice |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ voice |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ voice |= 0x000c;
++ break;
++ }
++
++ wm8753_write(codec, WM8753_PCM, voice);
++}
++
++/* configure PCM DAI */
++static int wm8753_pcm_dai_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 voice, ioctl, srate, srate2, fs, bfs, clock;
++ unsigned int rate;
++
++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
++ fs = rtd->codec_dai->dai_runtime.fs;
++ rate = snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate);
++ voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x001f;
++
++ /* set master/slave audio interface */
++ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x01fd;
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ ioctl |= 0x0002;
++ case SND_SOC_DAIFMT_CBM_CFS:
++ voice |= 0x0040;
++ break;
++ }
++
++ /* do we need to enable the PLL */
++ if (rtd->codec_dai->pll_in) {
++ if (wm8753_config_pll(codec, rtd->codec_dai, 2) !=
++ rtd->codec_dai->pll_in) {
++ err("could not set pll to %d --> %d",
++ rtd->codec_dai->pll_in, rtd->codec_dai->pll_out);
++ return -ENODEV;
++ }
++ }
++
++ /* set up PCM divider */
++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0x003f;
++ switch (rtd->codec_dai->clk_div) {
++ case 2: /* 1 */
++ break;
++ case 6: /* 3 */
++ clock |= (0x2 << 6);
++ break;
++ case 11: /* 5.5 */
++ clock |= (0x3 << 6);
++ break;
++ case 4: /* 2 */
++ clock |= (0x4 << 6);
++ break;
++ case 8: /* 4 */
++ clock |= (0x5 << 6);
++ break;
++ case 12: /* 6 */
++ clock |= (0x6 << 6);
++ break;
++ case 16: /* 8 */
++ clock |= (0x7 << 6);
++ break;
++ default:
++ printk(KERN_ERR "wm8753: invalid PCM clk divider %d\n",
++ rtd->codec_dai->clk_div);
++ break;
++ }
++ wm8753_write(codec, WM8753_CLOCK, clock);
++
++ /* set bclk divisor rate */
++ srate2 = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x003f;
++ switch (bfs) {
++ case 1:
++ break;
++ case 2:
++ srate2 |= (0x1 << 6);
++ break;
++ case 4:
++ srate2 |= (0x2 << 6);
++ break;
++ case 8:
++ srate2 |= (0x3 << 6);
++ break;
++ case 16:
++ srate2 |= (0x4 << 6);
++ break;
++ }
++ wm8753_write(codec, WM8753_SRATE2, srate2);
++
++ srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
++ if (rtd->codec_dai->dai_runtime.fs == 384)
++ srate |= 0x80;
++ wm8753_write(codec, WM8753_SRATE1, srate);
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_IB_IF:
++ voice |= 0x0090;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ voice |= 0x0080;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ voice |= 0x0010;
++ break;
++ }
++ //printk("voice %x %x ioctl %x %x srate2 %x %x srate1 %x %x\n",
++ //WM8753_PCM, voice, WM8753_IOCTL, ioctl, WM8753_SRATE2,
++ //srate2, WM8753_SRATE1, srate);
++
++ wm8753_write(codec, WM8753_IOCTL, ioctl);
++ wm8753_write(codec, WM8753_PCM, voice);
++ return 0;
++}
++
++/* configure hifi DAC wordlength and format */
++static void wm8753_hdac_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01e0;
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ hifi |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ hifi |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ hifi |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ hifi |= 0x0013;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ hifi |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ hifi |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ hifi |= 0x000c;
++ break;
++ }
++
++ wm8753_write(codec, WM8753_HIFI, hifi);
++}
++
++/* configure i2s (hifi) DAI clocking */
++static int wm8753_i2s_dai_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 srate, bfs, hifi, ioctl;
++ unsigned int rate;
++ int i = 0;
++
++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
++ rate = snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate);
++ hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x001f;
++
++ /* is coefficient valid ? */
++ if ((i = get_coeff(rtd->codec_dai->mclk, rate)) < 0)
++ return i;
++
++ srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
++ wm8753_write(codec, WM8753_SRATE1, srate | (coeff_div[i].sr << 1) |
++ coeff_div[i].usb);
++
++ /* do we need to enable the PLL */
++ if (rtd->codec_dai->pll_in) {
++ if (wm8753_config_pll(codec, rtd->codec_dai, 1) !=
++ rtd->codec_dai->pll_in) {
++ err("could not set pll to %d --> %d",
++ rtd->codec_dai->pll_in, rtd->codec_dai->pll_out);
++ return -ENODEV;
++ }
++ }
++
++ /* set bclk divisor rate */
++ srate = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x01c7;
++ switch (bfs) {
++ case 1:
++ break;
++ case 2:
++ srate |= (0x1 << 3);
++ break;
++ case 4:
++ srate |= (0x2 << 3);
++ break;
++ case 8:
++ srate |= (0x3 << 3);
++ break;
++ case 16:
++ srate |= (0x4 << 3);
++ break;
++ }
++ wm8753_write(codec, WM8753_SRATE2, srate);
++
++ /* set master/slave audio interface */
++ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x00fe;
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ ioctl |= 0x0001;
++ case SND_SOC_DAIFMT_CBM_CFS:
++ hifi |= 0x0040;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_IB_IF:
++ hifi |= 0x0090;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ hifi |= 0x0080;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ hifi |= 0x0010;
++ break;
++ }
++ wm8753_write(codec, WM8753_IOCTL, ioctl);
++ wm8753_write(codec, WM8753_HIFI, hifi);
++ return 0;
++}
++
++static int wm8753_mode1v_prepare (struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 clock;
++
++ /* set clk source as pcmclk */
++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++ wm8753_write(codec, WM8753_CLOCK, clock);
++
++ wm8753_adc_vdac_prepare(substream);
++ return wm8753_pcm_dai_prepare(substream);
++}
++
++static int wm8753_mode1h_prepare (struct snd_pcm_substream *substream)
++{
++ wm8753_hdac_prepare(substream);
++ return wm8753_i2s_dai_prepare(substream);
++}
++
++static int wm8753_mode2_prepare (struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 clock;
++
++ /* set clk source as pcmclk */
++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++ wm8753_write(codec, WM8753_CLOCK, clock);
++
++ wm8753_adc_vdac_prepare(substream);
++ return wm8753_i2s_dai_prepare(substream);
++}
++
++static int wm8753_mode3_prepare (struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 clock;
++
++ /* set clk source as mclk */
++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++ wm8753_write(codec, WM8753_CLOCK, clock | 0x4);
++
++ wm8753_hdac_prepare(substream);
++ wm8753_adc_vdac_prepare(substream);
++ return wm8753_i2s_dai_prepare(substream);
++}
++
++static int wm8753_mode4_prepare (struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 clock;
++
++ /* set clk source as mclk */
++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++ wm8753_write(codec, WM8753_CLOCK, clock | 0x4);
++
++ wm8753_hdac_prepare(substream);
++ wm8753_adc_vdac_prepare(substream);
++ return wm8753_i2s_dai_prepare(substream);
++}
++
++static int wm8753_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
++
++ /* the digital mute covers the HiFi and Voice DAC's on the WM8753.
++ * make sure we check if they are not both active when we mute */
++ if (mute && dai->id == 1) {
++ if (!wm8753_dai[WM8753_DAI_VOICE].playback.active ||
++ !wm8753_dai[WM8753_DAI_HIFI].playback.active)
++ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
++ } else {
++ if (mute)
++ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
++ else
++ wm8753_write(codec, WM8753_DAC, mute_reg);
++ }
++
++ return 0;
++}
++
++static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* set vmid to 50k and unmute dac */
++ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ /* set vmid to 5k for quick power up */
++ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* mute dac and set vmid to 500k, enable VREF */
++ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ wm8753_write(codec, WM8753_PWR1, 0x0001);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++/*
++ * The WM8753 supports upto 4 different and mutually exclusive DAI
++ * configurations. This gives 2 PCM's available for use, hifi and voice.
++ * NOTE: The Voice PCM cannot play or caputure audio to the CPU as it's DAI
++ * is connected between the wm8753 and a BT codec or GSM modem.
++ *
++ * 1. Voice over PCM DAI - HIFI DAC over HIFI DAI
++ * 2. Voice over HIFI DAI - HIFI disabled
++ * 3. Voice disabled - HIFI over HIFI
++ * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
++ */
++static const struct snd_soc_codec_dai wm8753_all_dai[] = {
++/* DAI HiFi mode 1 */
++{ .name = "WM8753 HiFi",
++ .id = 1,
++ .playback = {
++ .stream_name = "HiFi Playback",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = { /* dummy for fast DAI switching */
++ .stream_name = "HiFi Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm8753_config_i2s_sysclk,
++ .digital_mute = wm8753_mute,
++ .ops = {
++ .prepare = wm8753_mode1h_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8753_hifi_modes),
++ .mode = wm8753_hifi_modes,},
++},
++/* DAI Voice mode 1 */
++{ .name = "WM8753 Voice",
++ .id = 1,
++ .playback = {
++ .stream_name = "Voice Playback",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .capture = {
++ .stream_name = "Voice Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm8753_config_pcm_sysclk,
++ .digital_mute = wm8753_mute,
++ .ops = {
++ .prepare = wm8753_mode1v_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8753_voice_modes),
++ .mode = wm8753_voice_modes,},
++},
++/* DAI HiFi mode 2 - dummy */
++{ .name = "WM8753 HiFi",
++ .id = 2,
++},
++/* DAI Voice mode 2 */
++{ .name = "WM8753 Voice",
++ .id = 2,
++ .playback = {
++ .stream_name = "Voice Playback",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .capture = {
++ .stream_name = "Voice Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm8753_config_i2s_sysclk,
++ .digital_mute = wm8753_mute,
++ .ops = {
++ .prepare = wm8753_mode2_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8753_voice_modes),
++ .mode = wm8753_voice_modes,},
++},
++/* DAI HiFi mode 3 */
++{ .name = "WM8753 HiFi",
++ .id = 3,
++ .playback = {
++ .stream_name = "HiFi Playback",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .stream_name = "HiFi Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm8753_config_i2s_sysclk,
++ .digital_mute = wm8753_mute,
++ .ops = {
++ .prepare = wm8753_mode3_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8753_hifi_modes),
++ .mode = wm8753_hifi_modes,},
++},
++/* DAI Voice mode 3 - dummy */
++{ .name = "WM8753 Voice",
++ .id = 3,
++},
++/* DAI HiFi mode 4 */
++{ .name = "WM8753 HiFi",
++ .id = 4,
++ .playback = {
++ .stream_name = "HiFi Playback",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .stream_name = "HiFi Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm8753_config_i2s_sysclk,
++ .digital_mute = wm8753_mute,
++ .ops = {
++ .prepare = wm8753_mode4_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8753_mixed_modes),
++ .mode = wm8753_mixed_modes,},
++},
++/* DAI Voice mode 4 - dummy */
++{ .name = "WM8753 Voice",
++ .id = 4,
++},
++};
++
++struct snd_soc_codec_dai wm8753_dai[2];
++EXPORT_SYMBOL_GPL(wm8753_dai);
++
++static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
++{
++ if (mode < 4) {
++ wm8753_dai[0] = wm8753_all_dai[mode << 1];
++ wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
++ }
++}
++
++static void wm8753_work(void *data)
++{
++ struct snd_soc_codec *codec = (struct snd_soc_codec *)data;
++ wm8753_dapm_event(codec, codec->dapm_state);
++}
++
++static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm8753_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
++ if (i + 1 == WM8753_RESET)
++ continue;
++ data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++
++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* charge wm8753 caps */
++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
++ codec->dapm_state = SNDRV_CTL_POWER_D0;
++ queue_delayed_work(wm8753_workq, &wm8753_dapm_work,
++ msecs_to_jiffies(caps_charge));
++ }
++
++ return 0;
++}
++
++/*
++ * initialise the WM8753 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8753_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg, ret = 0;
++
++ codec->name = "WM8753";
++ codec->owner = THIS_MODULE;
++ codec->read = wm8753_read_reg_cache;
++ codec->write = wm8753_write;
++ codec->dapm_event = wm8753_dapm_event;
++ codec->dai = wm8753_dai;
++ codec->num_dai = 2;
++ codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
++
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8753_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, wm8753_reg,
++ sizeof(u16) * ARRAY_SIZE(wm8753_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8753_reg);
++ wm8753_set_dai_mode(codec, 0);
++
++ wm8753_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* charge output caps */
++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
++ codec->dapm_state = SNDRV_CTL_POWER_D3hot;
++ queue_delayed_work(wm8753_workq,
++ &wm8753_dapm_work, msecs_to_jiffies(caps_charge));
++
++ /* set the update bits */
++ reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
++ wm8753_write(codec, WM8753_LDAC, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_RDAC);
++ wm8753_write(codec, WM8753_RDAC, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V);
++ wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V);
++ wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V);
++ wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V);
++ wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_LINVOL);
++ wm8753_write(codec, WM8753_LINVOL, reg | 0x0100);
++ reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
++ wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
++
++ wm8753_add_controls(codec);
++ wm8753_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++static struct snd_soc_device *wm8753_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8753 2 wire address is determined by GPIO5
++ * state during powerup.
++ * low = 0x1a
++ * high = 0x1b
++ */
++#define I2C_DRIVERID_WM8753 0xfefe /* liam - need a proper id */
++
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver wm8753_i2c_driver;
++static struct i2c_client client_template;
++
++static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = wm8753_socdev;
++ struct wm8753_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL){
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if (ret < 0) {
++ err("failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = wm8753_init(socdev);
++ if (ret < 0) {
++ err("failed to initialise WM8753\n");
++ goto err;
++ }
++
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int wm8753_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec *codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int wm8753_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, wm8753_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8753_i2c_driver = {
++ .driver = {
++ .name = "WM8753 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_WM8753,
++ .attach_adapter = wm8753_i2c_attach,
++ .detach_client = wm8753_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "WM8753",
++ .driver = &wm8753_i2c_driver,
++};
++#endif
++
++static int wm8753_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct wm8753_setup_data *setup;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ info("WM8753 Audio Codec %s", WM8753_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++ wm8753_socdev = socdev;
++ INIT_WORK(&wm8753_dapm_work, wm8753_work, codec);
++ wm8753_workq = create_workqueue("wm8753");
++ if (wm8753_workq == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&wm8753_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++ return ret;
++}
++
++/* power down chip */
++static int wm8753_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec->control_data)
++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ if (wm8753_workq)
++ destroy_workqueue(wm8753_workq);
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&wm8753_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8753 = {
++ .probe = wm8753_probe,
++ .remove = wm8753_remove,
++ .suspend = wm8753_suspend,
++ .resume = wm8753_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
++
++MODULE_DESCRIPTION("ASoC WM8753 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8753.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8753.h
+@@ -0,0 +1,91 @@
++/*
++ * wm8753.h -- audio driver for WM8753
++ *
++ * Copyright 2003 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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.
++ *
++ */
++
++#ifndef _WM8753_H
++#define _WM8753_H
++
++/* WM8753 register space */
++
++#define WM8753_DAC 0x01
++#define WM8753_ADC 0x02
++#define WM8753_PCM 0x03
++#define WM8753_HIFI 0x04
++#define WM8753_IOCTL 0x05
++#define WM8753_SRATE1 0x06
++#define WM8753_SRATE2 0x07
++#define WM8753_LDAC 0x08
++#define WM8753_RDAC 0x09
++#define WM8753_BASS 0x0a
++#define WM8753_TREBLE 0x0b
++#define WM8753_ALC1 0x0c
++#define WM8753_ALC2 0x0d
++#define WM8753_ALC3 0x0e
++#define WM8753_NGATE 0x0f
++#define WM8753_LADC 0x10
++#define WM8753_RADC 0x11
++#define WM8753_ADCTL1 0x12
++#define WM8753_3D 0x13
++#define WM8753_PWR1 0x14
++#define WM8753_PWR2 0x15
++#define WM8753_PWR3 0x16
++#define WM8753_PWR4 0x17
++#define WM8753_ID 0x18
++#define WM8753_INTPOL 0x19
++#define WM8753_INTEN 0x1a
++#define WM8753_GPIO1 0x1b
++#define WM8753_GPIO2 0x1c
++#define WM8753_RESET 0x1f
++#define WM8753_RECMIX1 0x20
++#define WM8753_RECMIX2 0x21
++#define WM8753_LOUTM1 0x22
++#define WM8753_LOUTM2 0x23
++#define WM8753_ROUTM1 0x24
++#define WM8753_ROUTM2 0x25
++#define WM8753_MOUTM1 0x26
++#define WM8753_MOUTM2 0x27
++#define WM8753_LOUT1V 0x28
++#define WM8753_ROUT1V 0x29
++#define WM8753_LOUT2V 0x2a
++#define WM8753_ROUT2V 0x2b
++#define WM8753_MOUTV 0x2c
++#define WM8753_OUTCTL 0x2d
++#define WM8753_ADCIN 0x2e
++#define WM8753_INCTL1 0x2f
++#define WM8753_INCTL2 0x30
++#define WM8753_LINVOL 0x31
++#define WM8753_RINVOL 0x32
++#define WM8753_MICBIAS 0x33
++#define WM8753_CLOCK 0x34
++#define WM8753_PLL1CTL1 0x35
++#define WM8753_PLL1CTL2 0x36
++#define WM8753_PLL1CTL3 0x37
++#define WM8753_PLL1CTL4 0x38
++#define WM8753_PLL2CTL1 0x39
++#define WM8753_PLL2CTL2 0x3a
++#define WM8753_PLL2CTL3 0x3b
++#define WM8753_PLL2CTL4 0x3c
++#define WM8753_BIASCTL 0x3d
++#define WM8753_ADCTL2 0x3f
++
++struct wm8753_setup_data {
++ unsigned short i2c_address;
++};
++
++#define WM8753_DAI_HIFI 0
++#define WM8753_DAI_VOICE 1
++
++extern struct snd_soc_codec_dai wm8753_dai[2];
++extern struct snd_soc_codec_device soc_codec_dev_wm8753;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8772.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8772.c
+@@ -0,0 +1,806 @@
++/*
++ * wm8772.c -- WM8772 ALSA Soc Audio driver
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8772.h"
++
++#define AUDIO_NAME "WM8772"
++#define WM8772_VERSION "0.3"
++
++/*
++ * wm8772 register cache
++ * We can't read the WM8772 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8772_reg[] = {
++ 0x00ff, 0x00ff, 0x0120, 0x0000, /* 0 */
++ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /* 4 */
++ 0x00ff, 0x0000, 0x0080, 0x0040, /* 8 */
++ 0x0000
++};
++
++#define WM8772_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_IB_NF)
++
++#define WM8772_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM8772_PRATES \
++ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
++ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
++
++#define WM8772_CRATES \
++ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
++ SNDRV_PCM_RATE_96000)
++
++static struct snd_soc_dai_mode wm8772_modes[] = {
++ /* common codec frame and clock master modes */
++ /* 32k */
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 768,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 512,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 384,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 192,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 128,
++ .bfs = 64,
++ },
++
++ /* 44.1k */
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 768,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 512,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 384,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 192,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 128,
++ .bfs = 64,
++ },
++
++ /* 48k */
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 768,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 512,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 384,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 192,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 128,
++ .bfs = 64,
++ },
++
++ /* 96k */
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 384,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 256,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8772_DIR,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 192,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8772_DIR,
++ .pcmrate = SND_SOC_DAI_BFS_RATE,
++ .fs = 128,
++ .bfs = 64,
++ },
++
++ /* 192k */
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_192000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 192,
++ .bfs = 64,
++ },
++ {
++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_192000,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .fs = 128,
++ .bfs = 64,
++ },
++
++ /* slave mode */
++ {
++ .fmt = WM8772_DAIFMT,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = WM8772_PRATES,
++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++ {
++ .fmt = WM8772_DAIFMT,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = WM8772_CRATES,
++ .pcmdir = SND_SOC_DAIDIR_CAPTURE,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++/*
++ * read wm8772 register cache
++ */
++static inline unsigned int wm8772_read_reg_cache(struct snd_soc_codec * codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg > WM8772_CACHE_REGNUM)
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write wm8772 register cache
++ */
++static inline void wm8772_write_reg_cache(struct snd_soc_codec * codec,
++ unsigned int reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg > WM8772_CACHE_REGNUM)
++ return;
++ cache[reg] = value;
++}
++
++static int wm8772_write(struct snd_soc_codec * codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D9 WM8772 register offset
++ * D8...D0 register data
++ */
++ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++ data[1] = value & 0x00ff;
++
++ wm8772_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -1;
++}
++
++#define wm8772_reset(c) wm8772_write(c, WM8772_RESET, 0)
++
++/*
++ * WM8772 Controls
++ */
++static const char *wm8772_zero_flag[] = {"All Ch", "Ch 1", "Ch 2", "Ch3"};
++
++static const struct soc_enum wm8772_enum[] = {
++SOC_ENUM_SINGLE(WM8772_DACCTRL, 0, 4, wm8772_zero_flag),
++};
++
++static const struct snd_kcontrol_new wm8772_snd_controls[] = {
++
++SOC_SINGLE("Left1 Playback Volume", WM8772_LDAC1VOL, 0, 255, 0),
++SOC_SINGLE("Left2 Playback Volume", WM8772_LDAC2VOL, 0, 255, 0),
++SOC_SINGLE("Left3 Playback Volume", WM8772_LDAC3VOL, 0, 255, 0),
++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC1VOL, 0, 255, 0),
++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC2VOL, 0, 255, 0),
++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC3VOL, 0, 255, 0),
++SOC_SINGLE("Master Playback Volume", WM8772_MDACVOL, 0, 255, 0),
++
++SOC_SINGLE("Playback Switch", WM8772_DACCH, 0, 1, 0),
++SOC_SINGLE("Capture Switch", WM8772_ADCCTRL, 2, 1, 0),
++
++SOC_SINGLE("Demp1 Playback Switch", WM8772_DACCTRL, 6, 1, 0),
++SOC_SINGLE("Demp2 Playback Switch", WM8772_DACCTRL, 7, 1, 0),
++SOC_SINGLE("Demp3 Playback Switch", WM8772_DACCTRL, 8, 1, 0),
++
++SOC_SINGLE("Phase Invert 1 Switch", WM8772_IFACE, 6, 1, 0),
++SOC_SINGLE("Phase Invert 2 Switch", WM8772_IFACE, 7, 1, 0),
++SOC_SINGLE("Phase Invert 3 Switch", WM8772_IFACE, 8, 1, 0),
++
++SOC_SINGLE("Playback ZC Switch", WM8772_DACCTRL, 0, 1, 0),
++
++SOC_SINGLE("Capture High Pass Switch", WM8772_ADCCTRL, 3, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm8772_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm8772_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8772_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++ return 0;
++}
++
++/* valid wm8772 mclk frequencies */
++static const int freq_table[5][6] = {
++ {4096000, 6144000, 8192000, 12288000, 16384000, 24576000},
++ {5644800, 8467000, 11289600, 16934000, 22579200, 33868800},
++ {6144000, 9216000, 12288000, 18432000, 24576000, 36864000},
++ {12288000, 18432000, 24576000, 36864000, 0, 0},
++ {24576000, 36864000, 0, 0, 0},
++};
++
++static unsigned int check_freq(int rate, unsigned int freq)
++{
++ int i;
++
++ for(i = 0; i < 6; i++) {
++ if(freq == freq_table[i][rate])
++ return freq;
++ }
++ return 0;
++}
++
++static unsigned int wm8772_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ switch (info->rate){
++ case 32000:
++ dai->mclk = check_freq(0, clk);
++ break;
++ case 44100:
++ dai->mclk = check_freq(1, clk);
++ break;
++ case 48000:
++ dai->mclk = check_freq(2, clk);
++ break;
++ case 96000:
++ dai->mclk = check_freq(3, clk);
++ break;
++ case 192000:
++ dai->mclk = check_freq(4, clk);
++ break;
++ default:
++ dai->mclk = 0;
++ }
++ return dai->mclk;
++}
++
++static int wm8772_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0xffc0;
++ u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xfe1f;
++ u16 aiface = 0;
++ u16 aiface_ctrl = wm8772_read_reg_cache(codec, WM8772_ADCCTRL) & 0xfcff;
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++
++ /* set master/slave audio interface */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ diface_ctrl |= 0x0010;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ break;
++ }
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ diface |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ diface |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ diface |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ diface |= 0x0007;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FORMAT_S20_3LE:
++ diface |= 0x0010;
++ break;
++ case SNDRV_PCM_FORMAT_S24_3LE:
++ diface |= 0x0020;
++ break;
++ case SNDRV_PCM_FORMAT_S32_LE:
++ diface |= 0x0030;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ diface |= 0x0008;
++ break;
++ }
++
++ /* set rate */
++ switch (rtd->codec_dai->dai_runtime.fs) {
++ case 768:
++ diface_ctrl |= (0x5 << 6);
++ break;
++ case 512:
++ diface_ctrl |= (0x4 << 6);
++ break;
++ case 384:
++ diface_ctrl |= (0x3 << 6);
++ break;
++ case 256:
++ diface_ctrl |= (0x2 << 6);
++ break;
++ case 192:
++ diface_ctrl |= (0x1 << 6);
++ break;
++ }
++
++ wm8772_write(codec, WM8772_DACRATE, diface_ctrl);
++ wm8772_write(codec, WM8772_IFACE, diface);
++
++ } else {
++
++ /* set master/slave audio interface */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ aiface |= 0x0010;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ break;
++ }
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ aiface |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ aiface |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ aiface |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ aiface |= 0x0003;
++ aiface_ctrl |= 0x0010;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ aiface |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ aiface |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ aiface |= 0x000c;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ aiface_ctrl |= 0x0020;
++ break;
++ }
++
++ /* set rate */
++ switch (rtd->codec_dai->dai_runtime.fs) {
++ case 768:
++ aiface |= (0x5 << 5);
++ break;
++ case 512:
++ aiface |= (0x4 << 5);
++ break;
++ case 384:
++ aiface |= (0x3 << 5);
++ break;
++ case 256:
++ aiface |= (0x2 << 5);
++ break;
++ }
++
++ wm8772_write(codec, WM8772_ADCCTRL, aiface_ctrl);
++ wm8772_write(codec, WM8772_ADCRATE, aiface);
++ }
++
++ return 0;
++}
++
++static int wm8772_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 master = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xffe0;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* vref/mid, clk and osc on, dac unmute, active */
++ wm8772_write(codec, WM8772_DACRATE, master);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except vref/vmid, dac mute, inactive */
++ wm8772_write(codec, WM8772_DACRATE, master | 0x0f);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* everything off, dac mute, inactive */
++ wm8772_write(codec, WM8772_DACRATE, master | 0x1f);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai wm8772_dai = {
++ .name = "WM8772",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 6,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = wm8772_config_sysclk,
++ .ops = {
++ .prepare = wm8772_pcm_prepare,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8772_modes),
++ .mode = wm8772_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(wm8772_dai);
++
++static int wm8772_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm8772_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(wm8772_reg); i++) {
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ wm8772_dapm_event(codec, codec->suspend_dapm_state);
++ return 0;
++}
++
++/*
++ * initialise the WM8772 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8772_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg, ret = 0;
++
++ codec->name = "WM8772";
++ codec->owner = THIS_MODULE;
++ codec->read = wm8772_read_reg_cache;
++ codec->write = wm8772_write;
++ codec->dapm_event = wm8772_dapm_event;
++ codec->dai = &wm8772_dai;
++ codec->num_dai = 1;
++ codec->reg_cache_size = ARRAY_SIZE(wm8772_reg);
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8772_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, wm8772_reg,
++ sizeof(u16) * ARRAY_SIZE(wm8772_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8772_reg);
++
++ wm8772_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if(ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* power on device */
++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* set the update bits */
++ reg = wm8772_read_reg_cache(codec, WM8772_MDACVOL);
++ wm8772_write(codec, WM8772_MDACVOL, reg | 0x0100);
++ reg = wm8772_read_reg_cache(codec, WM8772_LDAC1VOL);
++ wm8772_write(codec, WM8772_LDAC1VOL, reg | 0x0100);
++ reg = wm8772_read_reg_cache(codec, WM8772_LDAC2VOL);
++ wm8772_write(codec, WM8772_LDAC2VOL, reg | 0x0100);
++ reg = wm8772_read_reg_cache(codec, WM8772_LDAC3VOL);
++ wm8772_write(codec, WM8772_LDAC3VOL, reg | 0x0100);
++ reg = wm8772_read_reg_cache(codec, WM8772_RDAC1VOL);
++ wm8772_write(codec, WM8772_RDAC1VOL, reg | 0x0100);
++ reg = wm8772_read_reg_cache(codec, WM8772_RDAC2VOL);
++ wm8772_write(codec, WM8772_RDAC2VOL, reg | 0x0100);
++ reg = wm8772_read_reg_cache(codec, WM8772_RDAC3VOL);
++ wm8772_write(codec, WM8772_RDAC3VOL, reg | 0x0100);
++
++ wm8772_add_controls(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++static struct snd_soc_device *wm8772_socdev;
++
++static int wm8772_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct wm8772_setup_data *setup;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ printk(KERN_INFO "WM8772 Audio Codec %s", WM8772_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ wm8772_socdev = socdev;
++
++ /* Add other interfaces here */
++#warning do SPI device probe here and then call wm8772_init()
++
++ return ret;
++}
++
++/* power down chip */
++static int wm8772_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec->control_data)
++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++ snd_soc_free_pcms(socdev);
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8772 = {
++ .probe = wm8772_probe,
++ .remove = wm8772_remove,
++ .suspend = wm8772_suspend,
++ .resume = wm8772_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8772);
++
++MODULE_DESCRIPTION("ASoC WM8772 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8772.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8772.h
+@@ -0,0 +1,40 @@
++/*
++ * wm8772.h -- audio driver for WM8772
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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.
++ *
++ */
++
++#ifndef _WM8772_H
++#define _WM8772_H
++
++/* WM8772 register space */
++
++#define WM8772_LDAC1VOL 0x00
++#define WM8772_RDAC1VOL 0x01
++#define WM8772_DACCH 0x02
++#define WM8772_IFACE 0x03
++#define WM8772_LDAC2VOL 0x04
++#define WM8772_RDAC2VOL 0x05
++#define WM8772_LDAC3VOL 0x06
++#define WM8772_RDAC3VOL 0x07
++#define WM8772_MDACVOL 0x08
++#define WM8772_DACCTRL 0x09
++#define WM8772_DACRATE 0x0a
++#define WM8772_ADCRATE 0x0b
++#define WM8772_ADCCTRL 0x0c
++#define WM8772_RESET 0x1f
++
++#define WM8772_CACHE_REGNUM 10
++
++extern struct snd_soc_codec_dai wm8772_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8772;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8971.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8971.c
+@@ -0,0 +1,1214 @@
++/*
++ * wm8971.c -- WM8971 ALSA SoC Audio driver
++ *
++ * Copyright 2005 Lab126, Inc.
++ *
++ * Author: Kenneth Kiraly <kiraly@lab126.com>
++ *
++ * Based on wm8753.c by Liam Girdwood
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8971.h"
++
++#define AUDIO_NAME "wm8971"
++#define WM8971_VERSION "0.8"
++
++#undef WM8971_DEBUG
++
++#ifdef WM8971_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++#define WM8971_REG_COUNT 43
++
++static struct workqueue_struct *wm8971_workq = NULL;
++static struct work_struct wm8971_dapm_work;
++
++/*
++ * wm8971 register cache
++ * We can't read the WM8971 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8971_reg[] = {
++ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
++ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
++ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
++ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
++ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
++ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
++ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
++ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
++ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
++ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
++ 0x0079, 0x0079, 0x0079, /* 40 */
++};
++
++#define WM8971_HIFI_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
++ SND_SOC_DAIFMT_IB_IF)
++
++#define WM8971_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM8971_HIFI_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++#define WM8971_HIFI_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8971_HIFI_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++static struct snd_soc_dai_mode wm8971_modes[] = {
++ /* common codec frame and clock master modes */
++ /* 8k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1408,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 2304,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 2112,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1500,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* 11.025k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1024,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1536,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1088,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* 16k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 768,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 1152,
++ .bfs = WM8971_HIFI_FSB
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 750,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* 22.05k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 512,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 768,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 544,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* 32k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 576,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 375,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* 44.1k & 48k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 384,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 272,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 250,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* 88.2k & 96k */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 128,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 192,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 136,
++ .bfs = WM8971_HIFI_FSB,
++ },
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 125,
++ .bfs = WM8971_HIFI_FSB,
++ },
++
++ /* codec frame and clock slave modes */
++ {
++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8971_HIFI_BITS,
++ .pcmrate = WM8971_HIFI_RATES,
++ .pcmdir = WM8971_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg < WM8971_REG_COUNT)
++ return cache[reg];
++
++ return -1;
++}
++
++static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg < WM8971_REG_COUNT)
++ cache[reg] = value;
++}
++
++static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D9 WM8753 register offset
++ * D8...D0 register data
++ */
++ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++ data[1] = value & 0x00ff;
++
++ wm8971_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -EIO;
++}
++
++#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0)
++
++/* WM8971 Controls */
++static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" };
++static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz",
++ "200Hz @ 48kHz" };
++static const char *wm8971_treble[] = { "8kHz", "4kHz" };
++static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" };
++static const char *wm8971_ng_type[] = { "Constant PGA Gain",
++ "Mute ADC Output" };
++static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
++static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)",
++ "Mono (Right)", "Digital Mono"};
++static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" };
++static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA",
++ "Differential"};
++static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA",
++ "Differential"};
++static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"};
++static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"};
++static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert",
++ "L + R Invert"};
++
++static const struct soc_enum wm8971_enum[] = {
++ SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */
++ SOC_ENUM_SINGLE(WM8971_BASS, 6, 2, wm8971_bass_filter),
++ SOC_ENUM_SINGLE(WM8971_TREBLE, 6, 2, wm8971_treble),
++ SOC_ENUM_SINGLE(WM8971_ALC1, 7, 4, wm8971_alc_func),
++ SOC_ENUM_SINGLE(WM8971_NGATE, 1, 2, wm8971_ng_type), /* 4 */
++ SOC_ENUM_SINGLE(WM8971_ADCDAC, 1, 4, wm8971_deemp),
++ SOC_ENUM_SINGLE(WM8971_ADCTL1, 4, 4, wm8971_mono_mux),
++ SOC_ENUM_SINGLE(WM8971_ADCTL1, 1, 2, wm8971_dac_phase),
++ SOC_ENUM_SINGLE(WM8971_LOUTM1, 0, 5, wm8971_lline_mux), /* 8 */
++ SOC_ENUM_SINGLE(WM8971_ROUTM1, 0, 5, wm8971_rline_mux),
++ SOC_ENUM_SINGLE(WM8971_LADCIN, 6, 4, wm8971_lpga_sel),
++ SOC_ENUM_SINGLE(WM8971_RADCIN, 6, 4, wm8971_rpga_sel),
++ SOC_ENUM_SINGLE(WM8971_ADCDAC, 5, 4, wm8971_adcpol), /* 12 */
++ SOC_ENUM_SINGLE(WM8971_ADCIN, 6, 4, wm8971_mono_mux),
++};
++
++static const struct snd_kcontrol_new wm8971_snd_controls[] = {
++ SOC_DOUBLE_R("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, 0, 63, 0),
++ SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL, 6, 1, 0),
++ SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1),
++
++ SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V,
++ WM8971_ROUT1V, 7, 1, 0),
++ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V,
++ WM8971_ROUT2V, 7, 1, 0),
++ SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0),
++
++ SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0),
++
++ SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1,
++ WM8971_LOUTM2, 4, 7, 1),
++ SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1,
++ WM8971_ROUTM2, 4, 7, 1),
++ SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1,
++ WM8971_MOUTM2, 4, 7, 1),
++
++ SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V,
++ WM8971_ROUT1V, 0, 127, 0),
++ SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V,
++ WM8971_ROUT2V, 0, 127, 0),
++
++ SOC_ENUM("Bass Boost", wm8971_enum[0]),
++ SOC_ENUM("Bass Filter", wm8971_enum[1]),
++ SOC_SINGLE("Bass Volume", WM8971_BASS, 0, 7, 1),
++
++ SOC_SINGLE("Treble Volume", WM8971_TREBLE, 0, 7, 0),
++ SOC_ENUM("Treble Cut-off", wm8971_enum[2]),
++
++ SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1),
++
++ SOC_SINGLE("ALC Target Volume", WM8971_ALC1, 0, 7, 0),
++ SOC_SINGLE("ALC Max Volume", WM8971_ALC1, 4, 7, 0),
++
++ SOC_SINGLE("ALC Capture Target Volume", WM8971_ALC1, 0, 7, 0),
++ SOC_SINGLE("ALC Capture Max Volume", WM8971_ALC1, 4, 7, 0),
++ SOC_ENUM("ALC Capture Function", wm8971_enum[3]),
++ SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0),
++ SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0),
++ SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0),
++ SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0),
++ SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0),
++ SOC_ENUM("ALC Capture NG Type", wm8971_enum[4]),
++ SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0),
++
++ SOC_SINGLE("Capture 6dB Attenuate", WM8971_ADCDAC, 8, 1, 0),
++ SOC_SINGLE("Playback 6dB Attenuate", WM8971_ADCDAC, 7, 1, 0),
++
++ SOC_ENUM("Playback De-emphasis", wm8971_enum[5]),
++ SOC_ENUM("Playback Function", wm8971_enum[6]),
++ SOC_ENUM("Playback Phase", wm8971_enum[7]),
++
++ SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
++};
++
++/* add non-DAPM controls */
++static int wm8971_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8971_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ return 0;
++}
++
++/*
++ * DAPM Controls
++ */
++
++/* Left Mixer */
++static const struct snd_kcontrol_new wm8971_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Playback Switch", WM8971_LOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_LOUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8971_LOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_LOUTM2, 7, 1, 0),
++};
++
++/* Right Mixer */
++static const struct snd_kcontrol_new wm8971_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8971_ROUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_ROUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Playback Switch", WM8971_ROUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_ROUTM2, 7, 1, 0),
++};
++
++/* Mono Mixer */
++static const struct snd_kcontrol_new wm8971_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8971_MOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_MOUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8971_MOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0),
++};
++
++/* Left Line Mux */
++static const struct snd_kcontrol_new wm8971_left_line_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[8]);
++
++/* Right Line Mux */
++static const struct snd_kcontrol_new wm8971_right_line_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[9]);
++
++/* Left PGA Mux */
++static const struct snd_kcontrol_new wm8971_left_pga_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[10]);
++
++/* Right PGA Mux */
++static const struct snd_kcontrol_new wm8971_right_pga_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[11]);
++
++/* Mono ADC Mux */
++static const struct snd_kcontrol_new wm8971_monomux_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[13]);
++
++static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = {
++ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8971_left_mixer_controls[0],
++ ARRAY_SIZE(wm8971_left_mixer_controls)),
++ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
++ &wm8971_right_mixer_controls[0],
++ ARRAY_SIZE(wm8971_right_mixer_controls)),
++ SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0,
++ &wm8971_mono_mixer_controls[0],
++ ARRAY_SIZE(wm8971_mono_mixer_controls)),
++
++ SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Right Out 1", WM8971_PWR2, 5, 0, NULL, 0),
++ SND_SOC_DAPM_PGA("Left Out 1", WM8971_PWR2, 6, 0, NULL, 0),
++ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8971_PWR2, 7, 0),
++ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0),
++ SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0),
++
++ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0),
++ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0),
++ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0),
++
++ SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0,
++ &wm8971_left_pga_controls),
++ SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0,
++ &wm8971_right_pga_controls),
++ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
++ &wm8971_left_line_controls),
++ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
++ &wm8971_right_line_controls),
++
++ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
++ &wm8971_monomux_controls),
++ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
++ &wm8971_monomux_controls),
++
++ SND_SOC_DAPM_OUTPUT("LOUT1"),
++ SND_SOC_DAPM_OUTPUT("ROUT1"),
++ SND_SOC_DAPM_OUTPUT("LOUT2"),
++ SND_SOC_DAPM_OUTPUT("ROUT2"),
++ SND_SOC_DAPM_OUTPUT("MONO"),
++
++ SND_SOC_DAPM_INPUT("LINPUT1"),
++ SND_SOC_DAPM_INPUT("RINPUT1"),
++ SND_SOC_DAPM_INPUT("MIC"),
++};
++
++static const char *audio_map[][3] = {
++ /* left mixer */
++ {"Left Mixer", "Playback Switch", "Left DAC"},
++ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
++ {"Left Mixer", "Right Playback Switch", "Right DAC"},
++ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
++
++ /* right mixer */
++ {"Right Mixer", "Left Playback Switch", "Left DAC"},
++ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
++ {"Right Mixer", "Playback Switch", "Right DAC"},
++ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
++
++ /* left out 1 */
++ {"Left Out 1", NULL, "Left Mixer"},
++ {"LOUT1", NULL, "Left Out 1"},
++
++ /* left out 2 */
++ {"Left Out 2", NULL, "Left Mixer"},
++ {"LOUT2", NULL, "Left Out 2"},
++
++ /* right out 1 */
++ {"Right Out 1", NULL, "Right Mixer"},
++ {"ROUT1", NULL, "Right Out 1"},
++
++ /* right out 2 */
++ {"Right Out 2", NULL, "Right Mixer"},
++ {"ROUT2", NULL, "Right Out 2"},
++
++ /* mono mixer */
++ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
++ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
++ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
++ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
++
++ /* mono out */
++ {"Mono Out", NULL, "Mono Mixer"},
++ {"MONO1", NULL, "Mono Out"},
++
++ /* Left Line Mux */
++ {"Left Line Mux", "Line", "LINPUT1"},
++ {"Left Line Mux", "PGA", "Left PGA Mux"},
++ {"Left Line Mux", "Differential", "Differential Mux"},
++
++ /* Right Line Mux */
++ {"Right Line Mux", "Line", "RINPUT1"},
++ {"Right Line Mux", "Mic", "MIC"},
++ {"Right Line Mux", "PGA", "Right PGA Mux"},
++ {"Right Line Mux", "Differential", "Differential Mux"},
++
++ /* Left PGA Mux */
++ {"Left PGA Mux", "Line", "LINPUT1"},
++ {"Left PGA Mux", "Differential", "Differential Mux"},
++
++ /* Right PGA Mux */
++ {"Right PGA Mux", "Line", "RINPUT1"},
++ {"Right PGA Mux", "Differential", "Differential Mux"},
++
++ /* Differential Mux */
++ {"Differential Mux", "Line", "LINPUT1"},
++ {"Differential Mux", "Line", "RINPUT1"},
++
++ /* Left ADC Mux */
++ {"Left ADC Mux", "Stereo", "Left PGA Mux"},
++ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
++ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
++
++ /* Right ADC Mux */
++ {"Right ADC Mux", "Stereo", "Right PGA Mux"},
++ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
++ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
++
++ /* ADC */
++ {"Left ADC", NULL, "Left ADC Mux"},
++ {"Right ADC", NULL, "Right ADC Mux"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int wm8971_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm8971_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8971_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++struct _coeff_div {
++ u32 mclk;
++ u32 rate;
++ u16 fs;
++ u8 sr:5;
++ u8 usb:1;
++};
++
++/* codec hifi mclk clock divider coefficients */
++static const struct _coeff_div coeff_div[] = {
++ /* 8k */
++ {12288000, 8000, 1536, 0x6, 0x0},
++ {11289600, 8000, 1408, 0x16, 0x0},
++ {18432000, 8000, 2304, 0x7, 0x0},
++ {16934400, 8000, 2112, 0x17, 0x0},
++ {12000000, 8000, 1500, 0x6, 0x1},
++
++ /* 11.025k */
++ {11289600, 11025, 1024, 0x18, 0x0},
++ {16934400, 11025, 1536, 0x19, 0x0},
++ {12000000, 11025, 1088, 0x19, 0x1},
++
++ /* 16k */
++ {12288000, 16000, 768, 0xa, 0x0},
++ {18432000, 16000, 1152, 0xb, 0x0},
++ {12000000, 16000, 750, 0xa, 0x1},
++
++ /* 22.05k */
++ {11289600, 22050, 512, 0x1a, 0x0},
++ {16934400, 22050, 768, 0x1b, 0x0},
++ {12000000, 22050, 544, 0x1b, 0x1},
++
++ /* 32k */
++ {12288000, 32000, 384, 0xc, 0x0},
++ {18432000, 32000, 576, 0xd, 0x0},
++ {12000000, 32000, 375, 0xa, 0x1},
++
++ /* 44.1k */
++ {11289600, 44100, 256, 0x10, 0x0},
++ {16934400, 44100, 384, 0x11, 0x0},
++ {12000000, 44100, 272, 0x11, 0x1},
++
++ /* 48k */
++ {12288000, 48000, 256, 0x0, 0x0},
++ {18432000, 48000, 384, 0x1, 0x0},
++ {12000000, 48000, 250, 0x0, 0x1},
++
++ /* 88.2k */
++ {11289600, 88200, 128, 0x1e, 0x0},
++ {16934400, 88200, 192, 0x1f, 0x0},
++ {12000000, 88200, 136, 0x1f, 0x1},
++
++ /* 96k */
++ {12288000, 96000, 128, 0xe, 0x0},
++ {18432000, 96000, 192, 0xf, 0x0},
++ {12000000, 96000, 125, 0xe, 0x1},
++};
++
++static int get_coeff(int mclk, int rate)
++{
++ int i;
++
++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
++ return i;
++ }
++ return -EINVAL;
++}
++
++/* WM8971 supports numerous input clocks per sample rate */
++static unsigned int wm8971_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ dai->mclk = 0;
++
++ /* check that the calculated FS and rate actually match a clock from
++ * the machine driver */
++ if (info->fs * info->rate == clk)
++ dai->mclk = clk;
++
++ return dai->mclk;
++}
++
++static int wm8971_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 iface = 0, bfs, srate = 0;
++ int i = get_coeff(rtd->codec_dai->mclk,
++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate));
++
++ /* is coefficient valid ? */
++ if (i < 0)
++ return i;
++
++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
++
++ /* set master/slave audio interface */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ iface |= 0x0040;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ break;
++ }
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ iface |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ iface |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ iface |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ iface |= 0x0013;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ iface |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ iface |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ iface |= 0x000c;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ break;
++ case SND_SOC_DAIFMT_IB_IF:
++ iface |= 0x0090;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ iface |= 0x0080;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ iface |= 0x0010;
++ break;
++ }
++
++ /* set bclk divisor rate */
++ switch (bfs) {
++ case 1:
++ break;
++ case 4:
++ srate |= (0x1 << 7);
++ break;
++ case 8:
++ srate |= (0x2 << 7);
++ break;
++ case 16:
++ srate |= (0x3 << 7);
++ break;
++ }
++
++ /* set iface & srate */
++ wm8971_write(codec, WM8971_AUDIO, iface);
++ wm8971_write(codec, WM8971_SRATE, srate |
++ (coeff_div[i].sr << 1) | coeff_div[i].usb);
++ return 0;
++}
++
++static int wm8971_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7;
++ if (mute)
++ wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8);
++ else
++ wm8971_write(codec, WM8971_ADCDAC, mute_reg);
++ return 0;
++}
++
++static int wm8971_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* set vmid to 50k and unmute dac */
++ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ /* set vmid to 5k for quick power up */
++ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x01c0);
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* mute dac and set vmid to 500k, enable VREF */
++ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ wm8971_write(codec, WM8971_PWR1, 0x0001);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai wm8971_dai = {
++ .name = "WM8971",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ },
++ .config_sysclk = wm8971_config_sysclk,
++ .digital_mute = wm8971_mute,
++ .ops = {
++ .prepare = wm8971_pcm_prepare,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8971_modes),
++ .mode = wm8971_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(wm8971_dai);
++
++static void wm8971_work(void *data)
++{
++ struct snd_soc_codec *codec = (struct snd_soc_codec *)data;
++ wm8971_dapm_event(codec, codec->dapm_state);
++}
++
++static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm8971_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) {
++ if (i + 1 == WM8971_RESET)
++ continue;
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++
++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* charge wm8971 caps */
++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2);
++ codec->dapm_state = SNDRV_CTL_POWER_D0;
++ queue_delayed_work(wm8971_workq, &wm8971_dapm_work,
++ msecs_to_jiffies(1000));
++ }
++
++ return 0;
++}
++
++static int wm8971_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg, ret = 0;
++
++ codec->name = "WM8971";
++ codec->owner = THIS_MODULE;
++ codec->read = wm8971_read_reg_cache;
++ codec->write = wm8971_write;
++ codec->dapm_event = wm8971_dapm_event;
++ codec->dai = &wm8971_dai;
++ codec->reg_cache_size = ARRAY_SIZE(wm8971_reg);
++ codec->num_dai = 1;
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8971_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, wm8971_reg,
++ sizeof(u16) * ARRAY_SIZE(wm8971_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8971_reg);
++
++ wm8971_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* charge output caps */
++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2);
++ codec->dapm_state = SNDRV_CTL_POWER_D3hot;
++ queue_delayed_work(wm8971_workq, &wm8971_dapm_work,
++ msecs_to_jiffies(1000));
++
++ /* set the update bits */
++ reg = wm8971_read_reg_cache(codec, WM8971_LDAC);
++ wm8971_write(codec, WM8971_LDAC, reg | 0x0100);
++ reg = wm8971_read_reg_cache(codec, WM8971_RDAC);
++ wm8971_write(codec, WM8971_RDAC, reg | 0x0100);
++
++ reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V);
++ wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100);
++ reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V);
++ wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100);
++
++ reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V);
++ wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100);
++ reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V);
++ wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100);
++
++ reg = wm8971_read_reg_cache(codec, WM8971_LINVOL);
++ wm8971_write(codec, WM8971_LINVOL, reg | 0x0100);
++ reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
++ wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
++
++ wm8971_add_controls(codec);
++ wm8971_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++static struct snd_soc_device *wm8971_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8731 2 wire address is determined by GPIO5
++ * state during powerup.
++ * low = 0x1a
++ * high = 0x1b
++ */
++#define I2C_DRIVERID_WM8971 0xfefe /* liam - need a proper id */
++
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver wm8971_i2c_driver;
++static struct i2c_client client_template;
++
++static int wm8971_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = wm8971_socdev;
++ struct wm8971_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++
++ i2c_set_clientdata(i2c, codec);
++
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if (ret < 0) {
++ err("failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = wm8971_init(socdev);
++ if (ret < 0) {
++ err("failed to initialise WM8971\n");
++ goto err;
++ }
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int wm8971_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec* codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int wm8971_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, wm8971_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8971_i2c_driver = {
++ .driver = {
++ .name = "WM8971 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_WM8971,
++ .attach_adapter = wm8971_i2c_attach,
++ .detach_client = wm8971_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "WM8971",
++ .driver = &wm8971_i2c_driver,
++};
++#endif
++
++static int wm8971_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct wm8971_setup_data *setup;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ info("WM8971 Audio Codec %s", WM8971_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++ wm8971_socdev = socdev;
++
++ INIT_WORK(&wm8971_dapm_work, wm8971_work, codec);
++ wm8971_workq = create_workqueue("wm8971");
++ if (wm8971_workq == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&wm8971_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++
++ return ret;
++}
++
++/* power down chip */
++static int wm8971_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec->control_data)
++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ if (wm8971_workq)
++ destroy_workqueue(wm8971_workq);
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&wm8971_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8971 = {
++ .probe = wm8971_probe,
++ .remove = wm8971_remove,
++ .suspend = wm8971_suspend,
++ .resume = wm8971_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
++
++MODULE_DESCRIPTION("ASoC WM8971 driver");
++MODULE_AUTHOR("Lab126");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8971.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8971.h
+@@ -0,0 +1,61 @@
++/*
++ * wm8971.h -- audio driver for WM8971
++ *
++ * Copyright 2005 Lab126, Inc.
++ *
++ * Author: Kenneth Kiraly <kiraly@lab126.com>
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ */
++
++#ifndef _WM8971_H
++#define _WM8971_H
++
++#define WM8971_LINVOL 0x00
++#define WM8971_RINVOL 0x01
++#define WM8971_LOUT1V 0x02
++#define WM8971_ROUT1V 0x03
++#define WM8971_ADCDAC 0x05
++#define WM8971_AUDIO 0x07
++#define WM8971_SRATE 0x08
++#define WM8971_LDAC 0x0a
++#define WM8971_RDAC 0x0b
++#define WM8971_BASS 0x0c
++#define WM8971_TREBLE 0x0d
++#define WM8971_RESET 0x0f
++#define WM8971_ALC1 0x11
++#define WM8971_ALC2 0x12
++#define WM8971_ALC3 0x13
++#define WM8971_NGATE 0x14
++#define WM8971_LADC 0x15
++#define WM8971_RADC 0x16
++#define WM8971_ADCTL1 0x17
++#define WM8971_ADCTL2 0x18
++#define WM8971_PWR1 0x19
++#define WM8971_PWR2 0x1a
++#define WM8971_ADCTL3 0x1b
++#define WM8971_ADCIN 0x1f
++#define WM8971_LADCIN 0x20
++#define WM8971_RADCIN 0x21
++#define WM8971_LOUTM1 0x22
++#define WM8971_LOUTM2 0x23
++#define WM8971_ROUTM1 0x24
++#define WM8971_ROUTM2 0x25
++#define WM8971_MOUTM1 0x26
++#define WM8971_MOUTM2 0x27
++#define WM8971_LOUT2V 0x28
++#define WM8971_ROUT2V 0x29
++#define WM8971_MOUTV 0x2A
++
++struct wm8971_setup_data {
++ unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8971_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8971;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8974.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8974.c
+@@ -0,0 +1,935 @@
++/*
++ * wm8974.c -- WM8974 ALSA Soc Audio driver
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ *
++ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8974.h"
++
++#define AUDIO_NAME "wm8974"
++#define WM8974_VERSION "0.5"
++
++/*
++ * Debug
++ */
++
++#define WM8974_DEBUG 0
++
++#ifdef WM8974_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++struct snd_soc_codec_device soc_codec_dev_wm8974;
++
++/*
++ * wm8974 register cache
++ * We can't read the WM8974 register space when we are
++ * using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0050, 0x0000, 0x0140, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x00ff,
++ 0x0000, 0x0000, 0x0100, 0x00ff,
++ 0x0000, 0x0000, 0x012c, 0x002c,
++ 0x002c, 0x002c, 0x002c, 0x0000,
++ 0x0032, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0038, 0x000b, 0x0032, 0x0000,
++ 0x0008, 0x000c, 0x0093, 0x00e9,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0003, 0x0010, 0x0000, 0x0000,
++ 0x0000, 0x0002, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0039, 0x0000,
++ 0x0000,
++};
++
++#define WM8974_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
++ SND_SOC_DAIFMT_IB_IF)
++
++#define WM8974_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM8974_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++#define WM8794_BCLK \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\
++ SND_SOC_FSBD(16) | SND_SOC_FSBD(32))
++
++#define WM8794_HIFI_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++static struct snd_soc_dai_mode wm8974_modes[] = {
++ /* codec frame and clock master modes */
++ {
++ .fmt = WM8974_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++ .pcmfmt = WM8794_HIFI_BITS,
++ .pcmrate = WM8974_RATES,
++ .pcmdir = WM8974_DIR,
++ .fs = 256,
++ .bfs = WM8794_BCLK,
++ },
++
++ /* codec frame and clock slave modes */
++ {
++ .fmt = WM8974_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM8794_HIFI_BITS,
++ .pcmrate = WM8974_RATES,
++ .pcmdir = WM8974_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++/*
++ * read wm8974 register cache
++ */
++static inline unsigned int wm8974_read_reg_cache(struct snd_soc_codec * codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg == WM8974_RESET)
++ return 0;
++ if (reg >= WM8974_CACHEREGNUM)
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write wm8974 register cache
++ */
++static inline void wm8974_write_reg_cache(struct snd_soc_codec *codec,
++ u16 reg, unsigned int value)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg >= WM8974_CACHEREGNUM)
++ return;
++ cache[reg] = value;
++}
++
++/*
++ * write to the WM8974 register space
++ */
++static int wm8974_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ u8 data[2];
++
++ /* data is
++ * D15..D9 WM8974 register offset
++ * D8...D0 register data
++ */
++ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++ data[1] = value & 0x00ff;
++
++ wm8974_write_reg_cache (codec, reg, value);
++ if (codec->hw_write(codec->control_data, data, 2) == 2)
++ return 0;
++ else
++ return -EIO;
++}
++
++#define wm8974_reset(c) wm8974_write(c, WM8974_RESET, 0)
++
++static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" };
++static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
++static const char *wm8974_eqmode[] = {"Capture", "Playback" };
++static const char *wm8974_bw[] = {"Narrow", "Wide" };
++static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
++static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
++static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
++static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
++static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
++static const char *wm8974_alc[] = {"ALC", "Limiter" };
++
++static const struct soc_enum wm8974_enum[] = {
++ SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */
++ SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */
++ SOC_ENUM_SINGLE(WM8974_DAC, 4, 4, wm8974_deemp),
++ SOC_ENUM_SINGLE(WM8974_EQ1, 8, 2, wm8974_eqmode),
++
++ SOC_ENUM_SINGLE(WM8974_EQ1, 5, 4, wm8974_eq1),
++ SOC_ENUM_SINGLE(WM8974_EQ2, 8, 2, wm8974_bw),
++ SOC_ENUM_SINGLE(WM8974_EQ2, 5, 4, wm8974_eq2),
++ SOC_ENUM_SINGLE(WM8974_EQ3, 8, 2, wm8974_bw),
++
++ SOC_ENUM_SINGLE(WM8974_EQ3, 5, 4, wm8974_eq3),
++ SOC_ENUM_SINGLE(WM8974_EQ4, 8, 2, wm8974_bw),
++ SOC_ENUM_SINGLE(WM8974_EQ4, 5, 4, wm8974_eq4),
++ SOC_ENUM_SINGLE(WM8974_EQ5, 8, 2, wm8974_bw),
++
++ SOC_ENUM_SINGLE(WM8974_EQ5, 5, 4, wm8974_eq5),
++ SOC_ENUM_SINGLE(WM8974_ALC3, 8, 2, wm8974_alc),
++};
++
++static const struct snd_kcontrol_new wm8974_snd_controls[] = {
++
++SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0),
++
++SOC_ENUM("DAC Companding", wm8974_enum[1]),
++SOC_ENUM("ADC Companding", wm8974_enum[0]),
++
++SOC_ENUM("Playback De-emphasis", wm8974_enum[2]),
++SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0),
++
++SOC_SINGLE("PCM Volume", WM8974_DACVOL, 0, 127, 0),
++
++SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0),
++SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0),
++SOC_SINGLE("ADC Inversion Switch", WM8974_COMP, 0, 1, 0),
++
++SOC_SINGLE("Capture Volume", WM8974_ADCVOL, 0, 127, 0),
++
++SOC_ENUM("Equaliser Function", wm8974_enum[3]),
++SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]),
++SOC_SINGLE("EQ1 Volume", WM8974_EQ1, 0, 31, 1),
++
++SOC_ENUM("Equaliser EQ2 Bandwith", wm8974_enum[5]),
++SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]),
++SOC_SINGLE("EQ2 Volume", WM8974_EQ2, 0, 31, 1),
++
++SOC_ENUM("Equaliser EQ3 Bandwith", wm8974_enum[7]),
++SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]),
++SOC_SINGLE("EQ3 Volume", WM8974_EQ3, 0, 31, 1),
++
++SOC_ENUM("Equaliser EQ4 Bandwith", wm8974_enum[9]),
++SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]),
++SOC_SINGLE("EQ4 Volume", WM8974_EQ4, 0, 31, 1),
++
++SOC_ENUM("Equaliser EQ5 Bandwith", wm8974_enum[11]),
++SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]),
++SOC_SINGLE("EQ5 Volume", WM8974_EQ5, 0, 31, 1),
++
++SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1, 8, 1, 0),
++SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1, 4, 15, 0),
++SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1, 0, 15, 0),
++
++SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2, 4, 7, 0),
++SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2, 0, 15, 0),
++
++SOC_SINGLE("ALC Enable Switch", WM8974_ALC1, 8, 1, 0),
++SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1, 3, 7, 0),
++SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1, 0, 7, 0),
++
++SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2, 8, 1, 0),
++SOC_SINGLE("ALC Capture Hold", WM8974_ALC2, 4, 7, 0),
++SOC_SINGLE("ALC Capture Target", WM8974_ALC2, 0, 15, 0),
++
++SOC_ENUM("ALC Capture Mode", wm8974_enum[13]),
++SOC_SINGLE("ALC Capture Decay", WM8974_ALC3, 4, 15, 0),
++SOC_SINGLE("ALC Capture Attack", WM8974_ALC3, 0, 15, 0),
++
++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE, 3, 1, 0),
++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE, 0, 7, 0),
++
++SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA, 7, 1, 0),
++SOC_SINGLE("Capture PGA Volume", WM8974_INPPGA, 0, 63, 0),
++
++SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL, 7, 1, 0),
++SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL, 6, 1, 1),
++SOC_SINGLE("Speaker Playback Volume", WM8974_SPKVOL, 0, 63, 0),
++
++SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0),
++SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm8974_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm8974_snd_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8974_snd_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ return 0;
++}
++
++/* Speaker Output Mixer */
++static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0),
++SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1),
++};
++
++/* Mono Output Mixer */
++static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0),
++SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 1),
++};
++
++/* AUX Input boost vol */
++static const struct snd_kcontrol_new wm8974_aux_boost_controls =
++SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0);
++
++/* Mic Input boost vol */
++static const struct snd_kcontrol_new wm8974_mic_boost_controls =
++SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0);
++
++/* Capture boost switch */
++static const struct snd_kcontrol_new wm8974_capture_boost_controls =
++SOC_DAPM_SINGLE("Capture Boost Switch", WM8974_INPPGA, 6, 1, 0);
++
++/* Aux In to PGA */
++static const struct snd_kcontrol_new wm8974_aux_capture_boost_controls =
++SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8974_INPPGA, 2, 1, 0);
++
++/* Mic P In to PGA */
++static const struct snd_kcontrol_new wm8974_micp_capture_boost_controls =
++SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8974_INPPGA, 0, 1, 0);
++
++/* Mic N In to PGA */
++static const struct snd_kcontrol_new wm8974_micn_capture_boost_controls =
++SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8974_INPPGA, 1, 1, 0);
++
++static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = {
++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0,
++ &wm8974_speaker_mixer_controls[0],
++ ARRAY_SIZE(wm8974_speaker_mixer_controls)),
++SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0,
++ &wm8974_mono_mixer_controls[0],
++ ARRAY_SIZE(wm8974_mono_mixer_controls)),
++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0),
++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER3, 0, 0),
++SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mic PGA", WM8974_POWER2, 2, 0, NULL, 0),
++
++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
++ &wm8974_aux_boost_controls, 1),
++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
++ &wm8974_mic_boost_controls, 1),
++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
++ &wm8974_capture_boost_controls),
++
++SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0, NULL, 0),
++
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0),
++
++SND_SOC_DAPM_INPUT("MICN"),
++SND_SOC_DAPM_INPUT("MICP"),
++SND_SOC_DAPM_INPUT("AUX"),
++SND_SOC_DAPM_OUTPUT("MONOOUT"),
++SND_SOC_DAPM_OUTPUT("SPKOUTP"),
++SND_SOC_DAPM_OUTPUT("SPKOUTN"),
++};
++
++static const char *audio_map[][3] = {
++ /* Mono output mixer */
++ {"Mono Mixer", "PCM Playback Switch", "DAC"},
++ {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
++ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
++
++ /* Speaker output mixer */
++ {"Speaker Mixer", "PCM Playback Switch", "DAC"},
++ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
++ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
++
++ /* Outputs */
++ {"Mono Out", NULL, "Mono Mixer"},
++ {"MONOOUT", NULL, "Mono Out"},
++ {"SpkN Out", NULL, "Speaker Mixer"},
++ {"SpkP Out", NULL, "Speaker Mixer"},
++ {"SPKOUTN", NULL, "SpkN Out"},
++ {"SPKOUTP", NULL, "SpkP Out"},
++
++ /* Boost Mixer */
++ {"Boost Mixer", NULL, "ADC"},
++ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"},
++ {"Aux Boost", "Aux Volume", "Boost Mixer"},
++ {"Capture Boost", "Capture Switch", "Boost Mixer"},
++ {"Mic Boost", "Mic Volume", "Boost Mixer"},
++
++ /* Inputs */
++ {"MICP", NULL, "Mic Boost"},
++ {"MICN", NULL, "Mic PGA"},
++ {"Mic PGA", NULL, "Capture Boost"},
++ {"AUX", NULL, "Aux Input"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++static int wm8974_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm8974_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8974_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++struct pll_ {
++ unsigned int in_hz, out_hz;
++ unsigned int pre:4; /* prescale - 1 */
++ unsigned int n:4;
++ unsigned int k;
++};
++
++struct pll_ pll[] = {
++ {12000000, 11289600, 0, 7, 0x86c220},
++ {12000000, 12288000, 0, 8, 0x3126e8},
++ {13000000, 11289600, 0, 6, 0xf28bd4},
++ {13000000, 12288000, 0, 7, 0x8fd525},
++ {12288000, 11289600, 0, 7, 0x59999a},
++ {11289600, 12288000, 0, 8, 0x80dee9},
++ /* liam - add more entries */
++};
++
++static int set_pll(struct snd_soc_codec *codec, unsigned int in,
++ unsigned int out)
++{
++ int i;
++ u16 reg;
++
++ if(out == 0) {
++ reg = wm8974_read_reg_cache(codec, WM8974_POWER1);
++ wm8974_write(codec, WM8974_POWER1, reg & 0x1df);
++ return 0;
++ }
++
++ for(i = 0; i < ARRAY_SIZE(pll); i++) {
++ if (in == pll[i].in_hz && out == pll[i].out_hz) {
++ wm8974_write(codec, WM8974_PLLN, (pll[i].pre << 4) | pll[i].n);
++ wm8974_write(codec, WM8974_PLLK1, pll[i].k >> 18);
++ wm8974_write(codec, WM8974_PLLK1, (pll[i].k >> 9) && 0x1ff);
++ wm8974_write(codec, WM8974_PLLK1, pll[i].k && 0x1ff);
++ reg = wm8974_read_reg_cache(codec, WM8974_POWER1);
++ wm8974_write(codec, WM8974_POWER1, reg | 0x020);
++ return 0;
++ }
++ }
++ return -EINVAL;
++}
++
++/* mclk dividers * 2 */
++static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24};
++
++/* we need 256FS to drive the DAC's and ADC's */
++static unsigned int wm8974_config_sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ int i, j, best_clk = info->fs * info->rate;
++
++ /* can we run at this clk without the PLL ? */
++ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) {
++ if ((best_clk >> 1) * mclk_div[i] == clk) {
++ dai->pll_in = 0;
++ dai->clk_div = mclk_div[i];
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++
++ /* now check for PLL support */
++ for (i = 0; i < ARRAY_SIZE(pll); i++) {
++ if (pll[i].in_hz == clk) {
++ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) {
++ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) {
++ dai->pll_in = clk;
++ dai->pll_out = pll[i].out_hz;
++ dai->clk_div = mclk_div[j];
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++ }
++ }
++
++ /* this clk is not supported */
++ return 0;
++}
++
++static int wm8974_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct snd_soc_codec_dai *dai = rtd->codec_dai;
++ u16 iface = 0, bfs, clk = 0, adn;
++ int fs = 48000 << 7, i;
++
++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
++ switch (bfs) {
++ case 2:
++ clk |= 0x1 << 2;
++ break;
++ case 4:
++ clk |= 0x2 << 2;
++ break;
++ case 8:
++ clk |= 0x3 << 2;
++ break;
++ case 16:
++ clk |= 0x4 << 2;
++ break;
++ case 32:
++ clk |= 0x5 << 2;
++ break;
++ }
++
++ /* set master/slave audio interface */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ clk |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ break;
++ }
++
++ /* interface format */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ iface |= 0x0010;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ iface |= 0x0008;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ iface |= 0x00018;
++ break;
++ }
++
++ /* bit size */
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ iface |= 0x0020;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ iface |= 0x0040;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ iface |= 0x0060;
++ break;
++ }
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ break;
++ case SND_SOC_DAIFMT_IB_IF:
++ iface |= 0x0180;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ iface |= 0x0100;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ iface |= 0x0080;
++ break;
++ }
++
++ /* filter coefficient */
++ adn = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x1f1;
++ switch (rtd->codec_dai->dai_runtime.pcmrate) {
++ case SNDRV_PCM_RATE_8000:
++ adn |= 0x5 << 1;
++ fs = 8000 << 7;
++ break;
++ case SNDRV_PCM_RATE_11025:
++ adn |= 0x4 << 1;
++ fs = 11025 << 7;
++ break;
++ case SNDRV_PCM_RATE_16000:
++ adn |= 0x3 << 1;
++ fs = 16000 << 7;
++ break;
++ case SNDRV_PCM_RATE_22050:
++ adn |= 0x2 << 1;
++ fs = 22050 << 7;
++ break;
++ case SNDRV_PCM_RATE_32000:
++ adn |= 0x1 << 1;
++ fs = 32000 << 7;
++ break;
++ case SNDRV_PCM_RATE_44100:
++ fs = 44100 << 7;
++ break;
++ }
++
++ /* do we need to enable the PLL */
++ if(dai->pll_in)
++ set_pll(codec, dai->pll_in, dai->pll_out);
++
++ /* divide the clock to 256 fs */
++ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) {
++ if (dai->clk_div == mclk_div[i]) {
++ clk |= i << 5;
++ clk &= 0xff;
++ goto set;
++ }
++ }
++
++set:
++ /* set iface */
++ wm8974_write(codec, WM8974_IFACE, iface);
++ wm8974_write(codec, WM8974_CLOCK, clk);
++
++ return 0;
++}
++
++static int wm8974_hw_free(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ set_pll(codec, 0, 0);
++ return 0;
++}
++
++static int wm8974_mute(struct snd_soc_codec *codec,
++ struct snd_soc_codec_dai *dai, int mute)
++{
++ u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
++ if(mute)
++ wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
++ else
++ wm8974_write(codec, WM8974_DAC, mute_reg);
++ return 0;
++}
++
++/* liam need to make this lower power with dapm */
++static int wm8974_dapm_event(struct snd_soc_codec *codec, int event)
++{
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* vref/mid, clk and osc on, dac unmute, active */
++ wm8974_write(codec, WM8974_POWER1, 0x1ff);
++ wm8974_write(codec, WM8974_POWER2, 0x1ff);
++ wm8974_write(codec, WM8974_POWER3, 0x1ff);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except vref/vmid, dac mute, inactive */
++
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* everything off, dac mute, inactive */
++ wm8974_write(codec, WM8974_POWER1, 0x0);
++ wm8974_write(codec, WM8974_POWER2, 0x0);
++ wm8974_write(codec, WM8974_POWER3, 0x0);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++struct snd_soc_codec_dai wm8974_dai = {
++ .name = "WM8974 HiFi",
++ .playback = {
++ .stream_name = "Playback",
++ .channels_min = 1,
++ .channels_max = 1,
++ },
++ .capture = {
++ .stream_name = "Capture",
++ .channels_min = 1,
++ .channels_max = 1,
++ },
++ .config_sysclk = wm8974_config_sysclk,
++ .digital_mute = wm8974_mute,
++ .ops = {
++ .prepare = wm8974_pcm_prepare,
++ .hw_free = wm8974_hw_free,
++ },
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm8974_modes),
++ .mode = wm8974_modes,
++ },
++};
++EXPORT_SYMBOL_GPL(wm8974_dai);
++
++static int wm8974_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm8974_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i;
++ u8 data[2];
++ u16 *cache = codec->reg_cache;
++
++ /* Sync reg_cache with the hardware */
++ for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) {
++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++ data[1] = cache[i] & 0x00ff;
++ codec->hw_write(codec->control_data, data, 2);
++ }
++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ wm8974_dapm_event(codec, codec->suspend_dapm_state);
++ return 0;
++}
++
++/*
++ * initialise the WM8974 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8974_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int ret = 0;
++
++ codec->name = "WM8974";
++ codec->owner = THIS_MODULE;
++ codec->read = wm8974_read_reg_cache;
++ codec->write = wm8974_write;
++ codec->dapm_event = wm8974_dapm_event;
++ codec->dai = &wm8974_dai;
++ codec->num_dai = 1;
++ codec->reg_cache_size = ARRAY_SIZE(wm8974_reg);
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8974_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, wm8974_reg,
++ sizeof(u16) * ARRAY_SIZE(wm8974_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8974_reg);
++
++ wm8974_reset(codec);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if(ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ /* power on device */
++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ wm8974_add_controls(codec);
++ wm8974_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if(ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++static struct snd_soc_device *wm8974_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8974 2 wire address is 0x1a
++ */
++#define I2C_DRIVERID_WM8974 0xfefe /* liam - need a proper id */
++
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver wm8974_i2c_driver;
++static struct i2c_client client_template;
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++
++static int wm8974_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = wm8974_socdev;
++ struct wm8974_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if(ret < 0) {
++ err("failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = wm8974_init(socdev);
++ if(ret < 0) {
++ err("failed to initialise WM8974\n");
++ goto err;
++ }
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int wm8974_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec *codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int wm8974_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, wm8974_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8974_i2c_driver = {
++ .driver = {
++ .name = "WM8974 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_WM8974,
++ .attach_adapter = wm8974_i2c_attach,
++ .detach_client = wm8974_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "WM8974",
++ .driver = &wm8974_i2c_driver,
++};
++#endif
++
++static int wm8974_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct wm8974_setup_data *setup;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ info("WM8974 Audio Codec %s", WM8974_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ wm8974_socdev = socdev;
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&wm8974_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++ return ret;
++}
++
++/* power down chip */
++static int wm8974_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec->control_data)
++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&wm8974_i2c_driver);
++#endif
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8974 = {
++ .probe = wm8974_probe,
++ .remove = wm8974_remove,
++ .suspend = wm8974_suspend,
++ .resume = wm8974_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8974);
++
++MODULE_DESCRIPTION("ASoC WM8974 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm8974.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm8974.h
+@@ -0,0 +1,64 @@
++/*
++ * wm8974.h -- WM8974 Soc Audio driver
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef _WM8974_H
++#define _WM8974_H
++
++/* WM8974 register space */
++
++#define WM8974_RESET 0x0
++#define WM8974_POWER1 0x1
++#define WM8974_POWER2 0x2
++#define WM8974_POWER3 0x3
++#define WM8974_IFACE 0x4
++#define WM8974_COMP 0x5
++#define WM8974_CLOCK 0x6
++#define WM8974_ADD 0x7
++#define WM8974_GPIO 0x8
++#define WM8974_DAC 0xa
++#define WM8974_DACVOL 0xb
++#define WM8974_ADC 0xe
++#define WM8974_ADCVOL 0xf
++#define WM8974_EQ1 0x12
++#define WM8974_EQ2 0x13
++#define WM8974_EQ3 0x14
++#define WM8974_EQ4 0x15
++#define WM8974_EQ5 0x16
++#define WM8974_DACLIM1 0x18
++#define WM8974_DACLIM2 0x19
++#define WM8974_NOTCH1 0x1b
++#define WM8974_NOTCH2 0x1c
++#define WM8974_NOTCH3 0x1d
++#define WM8974_NOTCH4 0x1e
++#define WM8974_ALC1 0x20
++#define WM8974_ALC2 0x21
++#define WM8974_ALC3 0x22
++#define WM8974_NGATE 0x23
++#define WM8974_PLLN 0x24
++#define WM8974_PLLK1 0x25
++#define WM8974_PLLK2 0x26
++#define WM8974_PLLK3 0x27
++#define WM8974_ATTEN 0x28
++#define WM8974_INPUT 0x2c
++#define WM8974_INPPGA 0x2d
++#define WM8974_ADCBOOST 0x2f
++#define WM8974_OUTPUT 0x31
++#define WM8974_SPKMIX 0x32
++#define WM8974_SPKVOL 0x36
++#define WM8974_MONOMIX 0x38
++
++#define WM8974_CACHEREGNUM 57
++
++struct wm8974_setup_data {
++ unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8974_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8974;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm9712.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm9712.c
+@@ -0,0 +1,781 @@
++/*
++ * wm9712.c -- ALSA Soc WM9712 codec support
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 4th Feb 2006 Initial version.
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/ac97_codec.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#define WM9712_VERSION "0.4"
++
++static unsigned int ac97_read(struct snd_soc_codec *codec,
++ unsigned int reg);
++static int ac97_write(struct snd_soc_codec *codec,
++ unsigned int reg, unsigned int val);
++
++#define AC97_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define AC97_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++/* may need to expand this */
++static struct snd_soc_dai_mode ac97_modes[] = {
++ {
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE,
++ .pcmrate = AC97_RATES,
++ .pcmdir = AC97_DIR,
++ },
++};
++
++/*
++ * WM9712 register cache
++ */
++static const u16 wm9712_reg[] = {
++ 0x6174, 0x8000, 0x8000, 0x8000, // 6
++ 0xf0f0, 0xaaa0, 0xc008, 0x6808, // e
++ 0xe808, 0xaaa0, 0xad00, 0x8000, // 16
++ 0xe808, 0x3000, 0x8000, 0x0000, // 1e
++ 0x0000, 0x0000, 0x0000, 0x000f, // 26
++ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
++ 0x0000, 0xbb80, 0x0000, 0x0000, // 36
++ 0x0000, 0x2000, 0x0000, 0x0000, // 3e
++ 0x0000, 0x0000, 0x0000, 0x0000, // 46
++ 0x0000, 0x0000, 0xf83e, 0xffff, // 4e
++ 0x0000, 0x0000, 0x0000, 0xf83e, // 56
++ 0x0008, 0x0000, 0x0000, 0x0000, // 5e
++ 0xb032, 0x3e00, 0x0000, 0x0000, // 66
++ 0x0000, 0x0000, 0x0000, 0x0000, // 6e
++ 0x0000, 0x0000, 0x0000, 0x0006, // 76
++ 0x0001, 0x0000, 0x574d, 0x4c12, // 7e
++ 0x0000, 0x0000 // virtual hp mixers
++};
++
++/* virtual HP mixers regs */
++#define HPL_MIXER 0x80
++#define HPR_MIXER 0x82
++
++static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
++static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
++static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
++ "Mono"};
++static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
++static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
++static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
++static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
++static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
++ "Stereo"};
++static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
++ "Line", "Headphone Mixer", "Phone Mixer", "Phone"};
++static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
++static const char *wm9712_diff_sel[] = {"Mic", "Line"};
++
++static const struct soc_enum wm9712_enum[] = {
++SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
++SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
++SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
++SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
++SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
++SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
++SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
++SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
++SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
++SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
++SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
++SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
++};
++
++static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
++SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
++SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
++SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
++SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
++
++SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
++SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
++SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
++SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
++SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0),
++
++SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
++SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
++SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
++SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
++SOC_ENUM("ALC Function", wm9712_enum[0]),
++SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
++SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
++SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
++SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
++SOC_ENUM("ALC NG Type", wm9712_enum[10]),
++SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
++
++SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1),
++SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
++
++SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
++SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
++SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
++
++SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
++SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
++SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
++
++SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
++SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
++SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
++
++SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0),
++SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
++
++SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
++SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
++
++SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
++SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
++SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
++
++SOC_ENUM("Bass Control", wm9712_enum[5]),
++SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
++SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
++SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
++SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0),
++SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0),
++
++SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
++SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
++SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
++SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
++
++SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
++SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
++SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm9712_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++ return 0;
++}
++
++/* We have to create a fake left and right HP mixers because
++ * the codec only has a single control that is shared by both channels.
++ * This makes it impossible to determine the audio path.
++ */
++static int mixer_event (struct snd_soc_dapm_widget *w, int event)
++{
++ u16 l, r, beep, line, phone, mic, pcm, aux;
++
++ l = ac97_read(w->codec, HPL_MIXER);
++ r = ac97_read(w->codec, HPR_MIXER);
++ beep = ac97_read(w->codec, AC97_PC_BEEP);
++ mic = ac97_read(w->codec, AC97_VIDEO);
++ phone = ac97_read(w->codec, AC97_PHONE);
++ line = ac97_read(w->codec, AC97_LINE);
++ pcm = ac97_read(w->codec, AC97_PCM);
++ aux = ac97_read(w->codec, AC97_CD);
++
++ if (l & 0x1 || r & 0x1)
++ ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
++
++ if (l & 0x2 || r & 0x2)
++ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
++
++ if (l & 0x4 || r & 0x4)
++ ac97_write(w->codec, AC97_LINE, line & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_LINE, line | 0x8000);
++
++ if (l & 0x8 || r & 0x8)
++ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
++
++ if (l & 0x10 || r & 0x10)
++ ac97_write(w->codec, AC97_CD, aux & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_CD, aux | 0x8000);
++
++ if (l & 0x20 || r & 0x20)
++ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
++
++ return 0;
++}
++
++/* Left Headphone Mixers */
++static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
++ SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
++ SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
++ SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
++ SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
++ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
++};
++
++/* Right Headphone Mixers */
++static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
++ SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
++ SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
++ SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
++ SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
++ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0),
++};
++
++/* Speaker Mixer */
++static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
++ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
++ SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
++ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
++ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
++};
++
++/* Phone Mixer */
++static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
++ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
++ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
++ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
++ SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
++ SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
++};
++
++/* ALC headphone mux */
++static const struct snd_kcontrol_new wm9712_alc_mux_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[1]);
++
++/* out 3 mux */
++static const struct snd_kcontrol_new wm9712_out3_mux_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[2]);
++
++/* spk mux */
++static const struct snd_kcontrol_new wm9712_spk_mux_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[3]);
++
++/* Capture to Phone mux */
++static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[4]);
++
++/* Capture left select */
++static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[8]);
++
++/* Capture right select */
++static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[9]);
++
++/* Mic select */
++static const struct snd_kcontrol_new wm9712_mic_src_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[7]);
++
++/* diff select */
++static const struct snd_kcontrol_new wm9712_diff_sel_controls =
++SOC_DAPM_ENUM("Route", wm9712_enum[11]);
++
++static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
++SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
++ &wm9712_alc_mux_controls),
++SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
++ &wm9712_out3_mux_controls),
++SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
++ &wm9712_spk_mux_controls),
++SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
++ &wm9712_capture_phone_mux_controls),
++SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
++ &wm9712_capture_selectl_controls),
++SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
++ &wm9712_capture_selectr_controls),
++SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
++ &wm9712_mic_src_controls),
++SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
++ &wm9712_diff_sel_controls),
++SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
++SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
++ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
++ mixer_event, SND_SOC_DAPM_POST_REG),
++SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
++ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
++ mixer_event, SND_SOC_DAPM_POST_REG),
++SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
++ &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
++SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
++ &wm9712_speaker_mixer_controls[0],
++ ARRAY_SIZE(wm9712_speaker_mixer_controls)),
++SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
++SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
++SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
++SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
++SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
++SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
++SND_SOC_DAPM_OUTPUT("MONOOUT"),
++SND_SOC_DAPM_OUTPUT("HPOUTL"),
++SND_SOC_DAPM_OUTPUT("HPOUTR"),
++SND_SOC_DAPM_OUTPUT("LOUT2"),
++SND_SOC_DAPM_OUTPUT("ROUT2"),
++SND_SOC_DAPM_OUTPUT("OUT3"),
++SND_SOC_DAPM_INPUT("LINEINL"),
++SND_SOC_DAPM_INPUT("LINEINR"),
++SND_SOC_DAPM_INPUT("PHONE"),
++SND_SOC_DAPM_INPUT("PCBEEP"),
++SND_SOC_DAPM_INPUT("MIC1"),
++SND_SOC_DAPM_INPUT("MIC2"),
++};
++
++static const char *audio_map[][3] = {
++ /* virtual mixer - mixes left & right channels for spk and mono */
++ {"AC97 Mixer", NULL, "Left DAC"},
++ {"AC97 Mixer", NULL, "Right DAC"},
++
++ /* Left HP mixer */
++ {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
++ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"},
++ {"Left HP Mixer", "Line Bypass Switch", "Line PGA"},
++ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
++ {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
++ {"Left HP Mixer", NULL, "ALC Sidetone Mux"},
++ //{"Right HP Mixer", NULL, "HP Mixer"},
++
++ /* Right HP mixer */
++ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
++ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"},
++ {"Right HP Mixer", "Line Bypass Switch", "Line PGA"},
++ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
++ {"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
++ {"Right HP Mixer", NULL, "ALC Sidetone Mux"},
++
++ /* speaker mixer */
++ {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
++ {"Speaker Mixer", "Line Bypass Switch", "Line PGA"},
++ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
++ {"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"},
++ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
++
++ /* Phone mixer */
++ {"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"},
++ {"Phone Mixer", "Line Bypass Switch", "Line PGA"},
++ {"Phone Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"},
++ {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
++ {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
++
++ /* inputs */
++ {"Line PGA", NULL, "LINEINL"},
++ {"Line PGA", NULL, "LINEINR"},
++ {"Phone PGA", NULL, "PHONE"},
++ {"Mic PGA", NULL, "MIC1"},
++ {"Mic PGA", NULL, "MIC2"},
++
++ /* left capture selector */
++ {"Left Capture Select", "Mic", "MIC1"},
++ {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
++ {"Left Capture Select", "Line", "LINEINL"},
++ {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
++ {"Left Capture Select", "Phone Mixer", "Phone Mixer"},
++ {"Left Capture Select", "Phone", "PHONE"},
++
++ /* right capture selector */
++ {"Right Capture Select", "Mic", "MIC2"},
++ {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
++ {"Right Capture Select", "Line", "LINEINR"},
++ {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
++ {"Right Capture Select", "Phone Mixer", "Phone Mixer"},
++ {"Right Capture Select", "Phone", "PHONE"},
++
++ /* ALC Sidetone */
++ {"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
++ {"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
++ {"ALC Sidetone Mux", "Left", "Left Capture Select"},
++ {"ALC Sidetone Mux", "Right", "Right Capture Select"},
++
++ /* ADC's */
++ {"Left ADC", NULL, "Left Capture Select"},
++ {"Right ADC", NULL, "Right Capture Select"},
++
++ /* outputs */
++ {"MONOOUT", NULL, "Phone Mixer"},
++ {"HPOUTL", NULL, "Headphone PGA"},
++ {"Headphone PGA", NULL, "Left HP Mixer"},
++ {"HPOUTR", NULL, "Headphone PGA"},
++ {"Headphone PGA", NULL, "Right HP Mixer"},
++
++ /* mono hp mixer */
++ {"Mono HP Mixer", NULL, "Left HP Mixer"},
++ {"Mono HP Mixer", NULL, "Right HP Mixer"},
++
++ /* Out3 Mux */
++ {"Out3 Mux", "Left", "Left HP Mixer"},
++ {"Out3 Mux", "Mono", "Phone Mixer"},
++ {"Out3 Mux", "Left + Right", "Mono HP Mixer"},
++ {"Out 3 PGA", NULL, "Out3 Mux"},
++ {"OUT3", NULL, "Out 3 PGA"},
++
++ /* speaker Mux */
++ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
++ {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"},
++ {"Speaker PGA", NULL, "Speaker Mux"},
++ {"LOUT2", NULL, "Speaker PGA"},
++ {"ROUT2", NULL, "Speaker PGA"},
++
++ {NULL, NULL, NULL},
++};
++
++static int wm9712_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++static unsigned int ac97_read(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++
++ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
++ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
++ reg == AC97_REC_GAIN)
++ return soc_ac97_ops.read(codec->ac97, reg);
++ else {
++ reg = reg >> 1;
++
++ if (reg > (ARRAY_SIZE(wm9712_reg)))
++ return -EIO;
++
++ return cache[reg];
++ }
++}
++
++static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int val)
++{
++ u16 *cache = codec->reg_cache;
++
++ soc_ac97_ops.write(codec->ac97, reg, val);
++ reg = reg >> 1;
++ if (reg <= (ARRAY_SIZE(wm9712_reg)))
++ cache[reg] = val;
++
++ return 0;
++}
++
++static int ac97_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg;
++ u16 vra;
++
++ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ reg = AC97_PCM_FRONT_DAC_RATE;
++ else
++ reg = AC97_PCM_LR_ADC_RATE;
++
++ return ac97_write(codec, reg, runtime->rate);
++}
++
++static int ac97_aux_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 vra, xsle;
++
++ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
++ xsle = ac97_read(codec, AC97_PCI_SID);
++ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
++
++ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
++ return -ENODEV;
++
++ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
++}
++
++struct snd_soc_codec_dai wm9712_dai[] = {
++{
++ .name = "AC97 HiFi",
++ .playback = {
++ .stream_name = "HiFi Playback",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .stream_name = "HiFi Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .ops = {
++ .prepare = ac97_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(ac97_modes),
++ .mode = ac97_modes,},
++ },
++ {
++ .name = "AC97 Aux",
++ .playback = {
++ .stream_name = "Aux Playback",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .ops = {
++ .prepare = ac97_aux_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(ac97_modes),
++ .mode = ac97_modes,},
++ },
++};
++EXPORT_SYMBOL_GPL(wm9712_dai);
++
++static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 reg;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* liam - maybe enable thermal shutdown */
++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff;
++ ac97_write(codec, AC97_EXTENDED_MID, reg);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* enable master bias and vmid */
++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff;
++ ac97_write(codec, AC97_EXTENDED_MID, reg);
++ ac97_write(codec, AC97_POWERDOWN, 0x0000);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* disable everything including AC link */
++ ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
++ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
++ ac97_write(codec, AC97_POWERDOWN, 0xffff);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
++{
++ if (try_warm && soc_ac97_ops.warm_reset) {
++ soc_ac97_ops.warm_reset(codec->ac97);
++ if (!(ac97_read(codec, 0) & 0x8000))
++ return 1;
++ }
++
++ soc_ac97_ops.reset(codec->ac97);
++ if (ac97_read(codec, 0) & 0x8000)
++ goto err;
++ return 0;
++
++err:
++ printk(KERN_ERR "WM9712 AC97 reset failed\n");
++ return -EIO;
++}
++
++static int wm9712_soc_suspend(struct platform_device *pdev,
++ pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm9712_soc_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ int i, ret;
++ u16 *cache = codec->reg_cache;
++
++ ret = wm9712_reset(codec, 1);
++ if (ret < 0){
++ printk(KERN_ERR "could not reset AC97 codec\n");
++ return ret;
++ }
++
++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ if (ret == 0) {
++ /* Sync reg_cache with the hardware after cold reset */
++ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
++ if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
++ (i > 0x58 && i != 0x5c))
++ continue;
++ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
++ }
++ }
++
++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
++
++ return ret;
++}
++
++static int wm9712_soc_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec;
++ int ret = 0;
++
++ printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
++
++ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (socdev->codec == NULL)
++ return -ENOMEM;
++ codec = socdev->codec;
++ mutex_init(&codec->mutex);
++
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9712_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL) {
++ kfree(codec->ac97);
++ kfree(socdev->codec);
++ socdev->codec = NULL;
++ return -ENOMEM;
++ }
++ memcpy(codec->reg_cache, wm9712_reg, sizeof(u16) * ARRAY_SIZE(wm9712_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9712_reg);
++ codec->reg_cache_step = 2;
++
++ codec->name = "WM9712";
++ codec->owner = THIS_MODULE;
++ codec->dai = wm9712_dai;
++ codec->num_dai = ARRAY_SIZE(wm9712_dai);
++ codec->write = ac97_write;
++ codec->read = ac97_read;
++ codec->dapm_event = wm9712_dapm_event;
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
++ if (ret < 0)
++ goto err;
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0)
++ goto pcm_err;
++
++ ret = wm9712_reset(codec, 0);
++ if (ret < 0) {
++ printk(KERN_ERR "AC97 link error\n");
++ goto reset_err;
++ }
++
++ /* set alc mux to none */
++ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
++
++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++ wm9712_add_controls(codec);
++ wm9712_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0)
++ goto reset_err;
++
++ return 0;
++
++reset_err:
++ snd_soc_free_pcms(socdev);
++
++pcm_err:
++ snd_soc_free_ac97_codec(codec);
++
++err:
++ kfree(socdev->codec->reg_cache);
++ kfree(socdev->codec);
++ socdev->codec = NULL;
++ return ret;
++}
++
++static int wm9712_soc_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec == NULL)
++ return 0;
++
++ snd_soc_dapm_free(socdev);
++ snd_soc_free_pcms(socdev);
++ snd_soc_free_ac97_codec(codec);
++ kfree(codec->reg_cache);
++ kfree(codec);
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm9712 = {
++ .probe = wm9712_soc_probe,
++ .remove = wm9712_soc_remove,
++ .suspend = wm9712_soc_suspend,
++ .resume = wm9712_soc_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
++
++MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm9712.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm9712.h
+@@ -0,0 +1,14 @@
++/*
++ * wm9712.h -- WM9712 Soc Audio driver
++ */
++
++#ifndef _WM9712_H
++#define _WM9712_H
++
++#define WM9712_DAI_AC97_HIFI 0
++#define WM9712_DAI_AC97_AUX 1
++
++extern struct snd_soc_codec_dai wm9712_dai[2];
++extern struct snd_soc_codec_device soc_codec_dev_wm9712;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm9713.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm9713.c
+@@ -0,0 +1,1313 @@
++/*
++ * wm9713.c -- ALSA Soc WM9713 codec support
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 4th Feb 2006 Initial version.
++ *
++ * Features:-
++ *
++ * o Support for AC97 Codec, Voice DAC and Aux DAC
++ * o Support for DAPM
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/ac97_codec.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#define WM9713_VERSION "0.12"
++
++struct wm9713 {
++ u32 pll; /* current PLL frequency */
++ u32 pll_resume; /* PLL resume frequency */
++};
++
++static unsigned int ac97_read(struct snd_soc_codec *codec,
++ unsigned int reg);
++static int ac97_write(struct snd_soc_codec *codec,
++ unsigned int reg, unsigned int val);
++
++#define AC97_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define AC97_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
++ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++/* may need to expand this */
++static struct snd_soc_dai_mode ac97_modes[] = {
++ {
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE,
++ .pcmrate = AC97_RATES,
++ },
++};
++
++#define WM9713_VOICE_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \
++ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_DSP_A | \
++ SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \
++ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
++ SND_SOC_DAIFMT_IB_IF)
++
++#define WM9713_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define WM9713_VOICE_FSB \
++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
++
++#define WM9713_VOICE_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 | \
++ SNDRV_PCM_RATE_96000)
++
++#define WM9713_HIFI_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++/*
++ * Voice modes
++ */
++static struct snd_soc_dai_mode wm9713_voice_modes[] = {
++ /* master modes */
++ {
++ .fmt = WM9713_VOICE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | \
++ SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = WM9713_HIFI_BITS,
++ .pcmrate = WM9713_VOICE_RATES,
++ .pcmdir = WM9713_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = WM9713_VOICE_FSB,
++ },
++
++ /* slave modes */
++ {
++ .fmt = WM9713_VOICE_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = WM9713_HIFI_BITS,
++ .pcmrate = WM9713_VOICE_RATES,
++ .pcmdir = WM9713_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++};
++
++/*
++ * WM9713 register cache
++ * Reg 0x3c bit 15 is used by touch driver.
++ */
++static const u16 wm9713_reg[] = {
++ 0x6174, 0x8080, 0x8080, 0x8080, // 6
++ 0xc880, 0xe808, 0xe808, 0x0808, // e
++ 0x00da, 0x8000, 0xd600, 0xaaa0, // 16
++ 0xaaa0, 0xaaa0, 0x0000, 0x0000, // 1e
++ 0x0f0f, 0x0040, 0x0000, 0x7f00, // 26
++ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
++ 0x0000, 0xbb80, 0x0000, 0x4523, // 36
++ 0x0000, 0x2000, 0x7eff, 0xffff, // 3e
++ 0x0000, 0x0000, 0x0080, 0x0000, // 46
++ 0x0000, 0x0000, 0xfffe, 0xffff, // 4e
++ 0x0000, 0x0000, 0x0000, 0xfffe, // 56
++ 0x4000, 0x0000, 0x0000, 0x0000, // 5e
++ 0xb032, 0x3e00, 0x0000, 0x0000, // 66
++ 0x0000, 0x0000, 0x0000, 0x0000, // 6e
++ 0x0000, 0x0000, 0x0000, 0x0006, // 76
++ 0x0001, 0x0000, 0x574d, 0x4c13, // 7e
++ 0x0000, 0x0000, 0x0000 // virtual hp & mic mixers
++};
++
++/* virtual HP mixers regs */
++#define HPL_MIXER 0x80
++#define HPR_MIXER 0x82
++#define MICB_MUX 0x82
++
++static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
++static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
++static const char *wm9713_rec_src[] =
++ {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker",
++ "Mono Out", "Zh"};
++static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
++static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
++static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv",
++ "Mono Vmid", "Inv Vmid"};
++static const char *wm9713_spk_pga[] =
++ {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid",
++ "Speaker Vmid", "Inv Vmid"};
++static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone",
++ "Headphone Vmid"};
++static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"};
++static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"};
++static const char *wm9713_dac_inv[] =
++ {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone",
++ "Headphone Mono", "NC", "Vmid"};
++static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"};
++static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"};
++static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"};
++static const char *wm9713_micb_select[] = {"MPB", "MPA"};
++
++static const struct soc_enum wm9713_enum[] = {
++SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */
++SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */
++SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */
++SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */
++SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/
++SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */
++SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */
++SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
++SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
++SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
++SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
++SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
++};
++
++static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
++SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
++SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
++SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
++SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE,15, 7, 1, 1),
++SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
++SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
++SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
++SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
++
++SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
++SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
++
++SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
++SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
++SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 63, 0),
++SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
++
++SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
++SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
++SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
++
++SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
++SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
++SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
++SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
++SOC_ENUM("ALC Function", wm9713_enum[6]),
++SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
++SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
++SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
++SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
++SOC_ENUM("ALC NG Type", wm9713_enum[17]),
++SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
++
++SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0),
++SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
++
++SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
++SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
++SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1),
++
++SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
++SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
++SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1),
++
++SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1),
++SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
++SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
++SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
++
++SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
++SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
++SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
++
++SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
++SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
++SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
++
++SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
++SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
++SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1),
++
++SOC_ENUM("Bass Control", wm9713_enum[16]),
++SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
++SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
++SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
++SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
++SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
++
++SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
++SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
++SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
++};
++
++/* add non dapm controls */
++static int wm9713_add_controls(struct snd_soc_codec *codec)
++{
++ int err, i;
++
++ for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm9713_snd_ac97_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++ return 0;
++}
++
++/* We have to create a fake left and right HP mixers because
++ * the codec only has a single control that is shared by both channels.
++ * This makes it impossible to determine the audio path using the current
++ * register map, thus we add a new (virtual) register to help determine the
++ * audio route within the device.
++ */
++static int mixer_event (struct snd_soc_dapm_widget *w, int event)
++{
++ u16 l, r, beep, tone, phone, rec, pcm, aux;
++
++ l = ac97_read(w->codec, HPL_MIXER);
++ r = ac97_read(w->codec, HPR_MIXER);
++ beep = ac97_read(w->codec, AC97_PC_BEEP);
++ tone = ac97_read(w->codec, AC97_MASTER_TONE);
++ phone = ac97_read(w->codec, AC97_PHONE);
++ rec = ac97_read(w->codec, AC97_REC_SEL);
++ pcm = ac97_read(w->codec, AC97_PCM);
++ aux = ac97_read(w->codec, AC97_AUX);
++
++ if (event & SND_SOC_DAPM_PRE_REG)
++ return 0;
++ if (l & 0x1 || r & 0x1)
++ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
++
++ if (l & 0x2 || r & 0x2)
++ ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000);
++
++ if (l & 0x4 || r & 0x4)
++ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
++
++ if (l & 0x8 || r & 0x8)
++ ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000);
++
++ if (l & 0x10 || r & 0x10)
++ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
++
++ if (l & 0x20 || r & 0x20)
++ ac97_write(w->codec, AC97_AUX, aux & 0x7fff);
++ else
++ ac97_write(w->codec, AC97_AUX, aux | 0x8000);
++
++ return 0;
++}
++
++/* Left Headphone Mixers */
++static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
++SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
++SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
++SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
++SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
++};
++
++/* Right Headphone Mixers */
++static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
++SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
++SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
++SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
++SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0),
++};
++
++/* headphone capture mux */
++static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[1]);
++
++/* headphone mic mux */
++static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[0]);
++
++/* Speaker Mixer */
++static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
++SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
++SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
++SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
++SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
++SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1),
++SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
++};
++
++/* Mono Mixer */
++static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
++SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
++SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
++SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
++SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1),
++SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1),
++SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1),
++SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1),
++};
++
++/* mono mic mux */
++static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[2]);
++
++/* mono output mux */
++static const struct snd_kcontrol_new wm9713_mono_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[7]);
++
++/* speaker left output mux */
++static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[8]);
++
++/* speaker right output mux */
++static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[9]);
++
++/* headphone left output mux */
++static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[10]);
++
++/* headphone right output mux */
++static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[11]);
++
++/* Out3 mux */
++static const struct snd_kcontrol_new wm9713_out3_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[12]);
++
++/* Out4 mux */
++static const struct snd_kcontrol_new wm9713_out4_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[13]);
++
++/* DAC inv mux 1 */
++static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[14]);
++
++/* DAC inv mux 2 */
++static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[15]);
++
++/* Capture source left */
++static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[3]);
++
++/* Capture source right */
++static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[4]);
++
++/* mic source */
++static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[18]);
++
++/* mic source B virtual control */
++static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls =
++SOC_DAPM_ENUM("Route", wm9713_enum[19]);
++
++static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = {
++SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_hp_rec_mux_controls),
++SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_hp_mic_mux_controls),
++SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_mono_mic_mux_controls),
++SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_mono_mux_controls),
++SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_hp_spkl_mux_controls),
++SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_hp_spkr_mux_controls),
++SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_hpl_out_mux_controls),
++SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_hpr_out_mux_controls),
++SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_out3_mux_controls),
++SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0,
++ &wm9713_out4_mux_controls),
++SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0,
++ &wm9713_dac_inv1_mux_controls),
++SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0,
++ &wm9713_dac_inv2_mux_controls),
++SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
++ &wm9713_rec_srcl_mux_controls),
++SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
++ &wm9713_rec_srcr_mux_controls),
++SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
++ &wm9713_mic_sel_mux_controls ),
++SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
++ &wm9713_micb_sel_mux_controls ),
++SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
++ &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls),
++ mixer_event, SND_SOC_DAPM_POST_REG),
++SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
++ &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls),
++ mixer_event, SND_SOC_DAPM_POST_REG),
++SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
++ &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
++SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
++ &wm9713_speaker_mixer_controls[0],
++ ARRAY_SIZE(wm9713_speaker_mixer_controls)),
++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1),
++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1),
++SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
++SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
++SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
++SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
++SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1),
++SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1),
++SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0),
++SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0),
++SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1),
++SND_SOC_DAPM_OUTPUT("MONO"),
++SND_SOC_DAPM_OUTPUT("HPL"),
++SND_SOC_DAPM_OUTPUT("HPR"),
++SND_SOC_DAPM_OUTPUT("SPKL"),
++SND_SOC_DAPM_OUTPUT("SPKR"),
++SND_SOC_DAPM_OUTPUT("OUT3"),
++SND_SOC_DAPM_OUTPUT("OUT4"),
++SND_SOC_DAPM_INPUT("LINEL"),
++SND_SOC_DAPM_INPUT("LINER"),
++SND_SOC_DAPM_INPUT("MONOIN"),
++SND_SOC_DAPM_INPUT("PCBEEP"),
++SND_SOC_DAPM_INPUT("MIC1"),
++SND_SOC_DAPM_INPUT("MIC2A"),
++SND_SOC_DAPM_INPUT("MIC2B"),
++SND_SOC_DAPM_VMID("VMID"),
++};
++
++static const char *audio_map[][3] = {
++ /* left HP mixer */
++ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
++ {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"},
++ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
++ {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"},
++ {"Left HP Mixer", NULL, "Capture Headphone Mux"},
++
++ /* right HP mixer */
++ {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
++ {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"},
++ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
++ {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"},
++ {"Right HP Mixer", NULL, "Capture Headphone Mux"},
++
++ /* virtual mixer - mixes left & right channels for spk and mono */
++ {"AC97 Mixer", NULL, "Left DAC"},
++ {"AC97 Mixer", NULL, "Right DAC"},
++ {"Line Mixer", NULL, "Right Line In"},
++ {"Line Mixer", NULL, "Left Line In"},
++ {"HP Mixer", NULL, "Left HP Mixer"},
++ {"HP Mixer", NULL, "Right HP Mixer"},
++ {"Capture Mixer", NULL, "Left Capture Source"},
++ {"Capture Mixer", NULL, "Right Capture Source"},
++
++ /* speaker mixer */
++ {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
++ {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"},
++ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
++ {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"},
++
++ /* mono mixer */
++ {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
++ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
++ {"Mono Mixer", "Aux Playback Switch", "Aux DAC"},
++ {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"},
++ {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"},
++ {"Mono Mixer", NULL, "Capture Mono Mux"},
++
++ /* DAC inv mux 1 */
++ {"DAC Inv Mux 1", "Mono", "Mono Mixer"},
++ {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"},
++ {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"},
++ {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"},
++ {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"},
++
++ /* DAC inv mux 2 */
++ {"DAC Inv Mux 2", "Mono", "Mono Mixer"},
++ {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"},
++ {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"},
++ {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"},
++ {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"},
++
++ /* headphone left mux */
++ {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"},
++
++ /* headphone right mux */
++ {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"},
++
++ /* speaker left mux */
++ {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"},
++ {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"},
++ {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"},
++
++ /* speaker right mux */
++ {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"},
++ {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"},
++ {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"},
++
++ /* mono mux */
++ {"Mono Out Mux", "Mono", "Mono Mixer"},
++ {"Mono Out Mux", "Inv", "DAC Inv Mux 1"},
++
++ /* out 3 mux */
++ {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"},
++
++ /* out 4 mux */
++ {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"},
++
++ /* output pga */
++ {"HPL", NULL, "Left Headphone"},
++ {"Left Headphone", NULL, "Left Headphone Out Mux"},
++ {"HPR", NULL, "Right Headphone"},
++ {"Right Headphone", NULL, "Right Headphone Out Mux"},
++ {"OUT3", NULL, "Out 3"},
++ {"Out 3", NULL, "Out 3 Mux"},
++ {"OUT4", NULL, "Out 4"},
++ {"Out 4", NULL, "Out 4 Mux"},
++ {"SPKL", NULL, "Left Speaker"},
++ {"Left Speaker", NULL, "Left Speaker Out Mux"},
++ {"SPKR", NULL, "Right Speaker"},
++ {"Right Speaker", NULL, "Right Speaker Out Mux"},
++ {"MONO", NULL, "Mono Out"},
++ {"Mono Out", NULL, "Mono Out Mux"},
++
++ /* input pga */
++ {"Left Line In", NULL, "LINEL"},
++ {"Right Line In", NULL, "LINER"},
++ {"Mono In", NULL, "MONOIN"},
++ {"Mic A PGA", NULL, "Mic A Pre Amp"},
++ {"Mic B PGA", NULL, "Mic B Pre Amp"},
++
++ /* left capture select */
++ {"Left Capture Source", "Mic 1", "Mic A Pre Amp"},
++ {"Left Capture Source", "Mic 2", "Mic B Pre Amp"},
++ {"Left Capture Source", "Line", "LINEL"},
++ {"Left Capture Source", "Mono In", "MONOIN"},
++ {"Left Capture Source", "Headphone", "Left HP Mixer"},
++ {"Left Capture Source", "Speaker", "Speaker Mixer"},
++ {"Left Capture Source", "Mono Out", "Mono Mixer"},
++
++ /* right capture select */
++ {"Right Capture Source", "Mic 1", "Mic A Pre Amp"},
++ {"Right Capture Source", "Mic 2", "Mic B Pre Amp"},
++ {"Right Capture Source", "Line", "LINER"},
++ {"Right Capture Source", "Mono In", "MONOIN"},
++ {"Right Capture Source", "Headphone", "Right HP Mixer"},
++ {"Right Capture Source", "Speaker", "Speaker Mixer"},
++ {"Right Capture Source", "Mono Out", "Mono Mixer"},
++
++ /* left ADC */
++ {"Left ADC", NULL, "Left Capture Source"},
++
++ /* right ADC */
++ {"Right ADC", NULL, "Right Capture Source"},
++
++ /* mic */
++ {"Mic A Pre Amp", NULL, "Mic A Source"},
++ {"Mic A Source", "Mic 1", "MIC1"},
++ {"Mic A Source", "Mic 2 A", "MIC2A"},
++ {"Mic A Source", "Mic 2 B", "Mic B Source"},
++ {"Mic B Pre Amp", "MPB", "Mic B Source"},
++ {"Mic B Source", NULL, "MIC2B"},
++
++ /* headphone capture */
++ {"Capture Headphone Mux", "Stereo", "Capture Mixer"},
++ {"Capture Headphone Mux", "Left", "Left Capture Source"},
++ {"Capture Headphone Mux", "Right", "Right Capture Source"},
++
++ /* mono capture */
++ {"Capture Mono Mux", "Stereo", "Capture Mixer"},
++ {"Capture Mono Mux", "Left", "Left Capture Source"},
++ {"Capture Mono Mux", "Right", "Right Capture Source"},
++
++ {NULL, NULL, NULL},
++};
++
++static int wm9713_add_widgets(struct snd_soc_codec *codec)
++{
++ int i;
++
++ for(i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
++ }
++
++ /* set up audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_new_widgets(codec);
++ return 0;
++}
++
++static unsigned int ac97_read(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u16 *cache = codec->reg_cache;
++
++ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
++ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
++ reg == AC97_CD)
++ return soc_ac97_ops.read(codec->ac97, reg);
++ else {
++ reg = reg >> 1;
++
++ if (reg > (ARRAY_SIZE(wm9713_reg)))
++ return -EIO;
++
++ return cache[reg];
++ }
++}
++
++static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int val)
++{
++ u16 *cache = codec->reg_cache;
++ if (reg < 0x7c)
++ soc_ac97_ops.write(codec->ac97, reg, val);
++ reg = reg >> 1;
++ if (reg <= (ARRAY_SIZE(wm9713_reg)))
++ cache[reg] = val;
++
++ return 0;
++}
++
++struct pll_ {
++ unsigned int in_hz;
++ unsigned int lf:1; /* allows low frequency use */
++ unsigned int sdm:1; /* allows fraction n div */
++ unsigned int divsel:1; /* enables input clock div */
++ unsigned int divctl:1; /* input clock divider */
++ unsigned int n:4;
++ unsigned int k;
++};
++
++struct pll_ pll[] = {
++ {13000000, 0, 1, 0, 0, 7, 0x23f488},
++ {2048000, 1, 0, 0, 0, 12, 0x0},
++ {4096000, 1, 0, 0, 0, 6, 0x0},
++ {12288000, 0, 0, 0, 0, 8, 0x0},
++ /* liam - add more entries */
++};
++
++/* we must have either 24.576MHz or a PLL freq */
++static unsigned int wm9713_config_ac97sysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ int i;
++ dai->mclk = 0;
++
++ /* first check if we can get away witout burning any PLL power */
++ if (24576000 == clk) {
++ /* standard AC97 clock */
++ dai->mclk = clk;
++ goto out;
++ }
++
++ /* ok no standard clock, so we must now try the PLL */
++ for(i = 0; i < ARRAY_SIZE(pll); i++) {
++ if (clk == pll[i].in_hz) {
++ dai->mclk = clk; /* clock out */
++ goto out;
++ }
++ }
++
++out:
++ return dai->mclk;
++}
++
++/* The WM9713 voice DAC can only run at 256FS. This interface and DAC are
++ * clocked by the main AC97 clock divided down to 256 FS.
++ */
++static unsigned int wm9713_config_vsysclk(struct snd_soc_codec_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++
++ int i, j, best_clk = info->fs * info->rate;
++
++ /* can we run at this clk without the PLL ? */
++ for (i = 1; i <= 16; i++) {
++ if (best_clk * i == clk) {
++ dai->pll_in = 0;
++ dai->clk_div = i << 1;
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++
++ /* now check for PLL support */
++ for (i = 0; i < ARRAY_SIZE(pll); i++) {
++ if (pll[i].in_hz == clk) {
++ for (j = 1; j <= 16; j++) {
++ if (24576000 == j * best_clk) {
++ dai->pll_in = clk;
++ dai->pll_out = 24576000;
++ dai->clk_div = j << 1;
++ dai->mclk = best_clk;
++ return dai->mclk;
++ }
++ }
++ }
++ }
++
++ /* this clk is not supported */
++ return 0;
++}
++
++u32 wm9713_set_pll(struct snd_soc_codec *codec, u32 in)
++{
++ struct wm9713 *wm = (struct wm9713*)codec->private_data;
++ int i;
++ u16 reg, reg2;
++
++ /* turn PLL off ? */
++ if (in == 0) {
++ /* disable PLL power and select ext source */
++ reg = ac97_read(codec, AC97_HANDSET_RATE);
++ ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
++ reg = ac97_read(codec, AC97_EXTENDED_MID);
++ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
++ wm->pll = 0;
++ return 0;
++ }
++
++ for (i = 0; i < ARRAY_SIZE(pll); i++) {
++ if (pll[i].in_hz == in)
++ goto found;
++ }
++ return -EINVAL;
++
++found:
++ if (pll[i].sdm == 0) {
++ reg = (pll[i].n << 12) | (pll[i].lf << 11) |
++ (pll[i].divsel << 9) | (pll[i].divctl << 8);
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++ } else {
++ /* write the fractional k to the reg 0x46 pages */
++ reg2 = (pll[i].n << 12) | (pll[i].lf << 11) | (pll[i].sdm << 10) |
++ (pll[i].divsel << 9) | (pll[i].divctl << 8);
++
++ reg = reg2 | (0x5 << 4) | (pll[i].k >> 20); /* K [21:20] */
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++
++ reg = reg2 | (0x4 << 4) | ((pll[i].k >> 16) & 0xf); /* K [19:16] */
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++
++ reg = reg2 | (0x3 << 4) | ((pll[i].k >> 12) & 0xf); /* K [15:12] */
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++
++ reg = reg2 | (0x2 << 4) | ((pll[i].k >> 8) & 0xf); /* K [11:8] */
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++
++ reg = reg2 | (0x1 << 4) | ((pll[i].k >> 4) & 0xf); /* K [7:4] */
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++
++ reg = reg2 | (0x0 << 4) | (pll[i].k & 0xf); /* K [3:0] */
++ ac97_write(codec, AC97_LINE1_LEVEL, reg);
++ }
++
++ /* turn PLL on and select as source */
++ reg = ac97_read(codec, AC97_EXTENDED_MID);
++ ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
++ reg = ac97_read(codec, AC97_HANDSET_RATE);
++ ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
++ /* wait 10ms AC97 link frames for the link to stabilise */
++ schedule_timeout_interruptible(msecs_to_jiffies(10));
++ wm->pll = in;
++ return 0;
++}
++EXPORT_SYMBOL_GPL(wm9713_set_pll);
++
++static int wm9713_voice_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 reg = 0x8000, bfs, div, gpio;
++
++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
++ gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffe2;
++
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){
++ case SND_SOC_DAIFMT_CBM_CFM:
++ reg |= 0x4000;
++ gpio |= 0x0008;
++ break;
++ case SND_SOC_DAIFMT_CBM_CFS:
++ reg |= 0x6000;
++ gpio |= 0x000c;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFS:
++ reg |= 0x0200;
++ gpio |= 0x000d;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFM:
++ gpio |= 0x0009;
++ break;
++ }
++ ac97_write(codec, AC97_GPIO_CFG, gpio);
++
++ /* enable PLL if needed */
++ if (rtd->codec_dai->pll_in)
++ wm9713_set_pll(codec, rtd->codec_dai->pll_in);
++
++ /* set the PCM divider */
++ div = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff;
++ ac97_write(codec, AC97_HANDSET_RATE, div |
++ ((rtd->codec_dai->clk_div >> 1) -1) << 8);
++
++ /* clock inversion */
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_IB_IF:
++ reg |= 0x00c0;
++ break;
++ case SND_SOC_DAIFMT_IB_NF:
++ reg |= 0x0080;
++ break;
++ case SND_SOC_DAIFMT_NB_IF:
++ reg |= 0x0040;
++ break;
++ }
++
++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ case SND_SOC_DAIFMT_I2S:
++ reg |= 0x0002;
++ break;
++ case SND_SOC_DAIFMT_RIGHT_J:
++ break;
++ case SND_SOC_DAIFMT_LEFT_J:
++ reg |= 0x0001;
++ break;
++ case SND_SOC_DAIFMT_DSP_A:
++ reg |= 0x0003;
++ break;
++ case SND_SOC_DAIFMT_DSP_B:
++ reg |= 0x0043;
++ break;
++ }
++
++ switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ break;
++ case SNDRV_PCM_FMTBIT_S20_3LE:
++ reg |= 0x0004;
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ reg |= 0x0008;
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ reg |= 0x000c;
++ break;
++ }
++
++ switch (bfs) {
++ case 2:
++ reg |= (0x1 << 9);
++ break;
++ case 4:
++ reg |= (0x2 << 9);
++ break;
++ case 8:
++ reg |= (0x3 << 9);
++ break;
++ case 16:
++ reg |= (0x4 << 9);
++ break;
++ }
++
++ /* enable PCM interface in master mode */
++ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
++ return 0;
++}
++
++static void wm9713_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (!codec->active)
++ wm9713_set_pll(codec, 0);
++}
++
++static void wm9713_voiceshutdown(snd_pcm_substream_t *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 status;
++
++ wm9713_shutdown(substream);
++
++ /* Gracefully shut down the voice interface. */
++ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
++ ac97_write(codec,AC97_HANDSET_RATE,0x0280);
++ schedule_timeout_interruptible(msecs_to_jiffies(1));
++ ac97_write(codec,AC97_HANDSET_RATE,0x0F80);
++ ac97_write(codec,AC97_EXTENDED_MID,status);
++}
++
++static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ int reg;
++ u16 vra;
++
++ /* we need a 24576000Hz clock to run at the correct speed */
++ if (rtd->codec_dai->mclk != 24576000)
++ wm9713_set_pll(codec, rtd->codec_dai->mclk);
++
++ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ reg = AC97_PCM_FRONT_DAC_RATE;
++ else
++ reg = AC97_PCM_LR_ADC_RATE;
++
++ return ac97_write(codec, reg, runtime->rate);
++}
++
++static int ac97_aux_prepare(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_device *socdev = rtd->socdev;
++ struct snd_soc_codec *codec = socdev->codec;
++ u16 vra, xsle;
++
++ /* we need a 24576000Hz clock to run at the correct speed */
++ if (rtd->codec_dai->mclk != 24576000)
++ wm9713_set_pll(codec, rtd->codec_dai->mclk);
++
++ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
++ xsle = ac97_read(codec, AC97_PCI_SID);
++ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
++
++ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
++ return -ENODEV;
++
++ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
++}
++
++struct snd_soc_codec_dai wm9713_dai[] = {
++{
++ .name = "AC97 HiFi",
++ .playback = {
++ .stream_name = "HiFi Playback",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .stream_name = "HiFi Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm9713_config_ac97sysclk,
++ .ops = {
++ .shutdown = wm9713_shutdown,
++ .prepare = ac97_hifi_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(ac97_modes),
++ .mode = ac97_modes,},},
++ {
++ .name = "AC97 Aux",
++ .playback = {
++ .stream_name = "Aux Playback",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .config_sysclk = wm9713_config_ac97sysclk,
++ .ops = {
++ .shutdown = wm9713_shutdown,
++ .prepare = ac97_aux_prepare,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(ac97_modes),
++ .mode = ac97_modes,}
++ },
++ {
++ .name = "WM9713 Voice",
++ .playback = {
++ .stream_name = "Voice Playback",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .capture = {
++ .stream_name = "Voice Capture",
++ .channels_min = 1,
++ .channels_max = 2,},
++ .config_sysclk = wm9713_config_vsysclk,
++ .ops = {
++ .prepare = wm9713_voice_prepare,
++ .shutdown = wm9713_voiceshutdown,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(wm9713_voice_modes),
++ .mode = wm9713_voice_modes,},
++ },
++};
++EXPORT_SYMBOL_GPL(wm9713_dai);
++
++int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
++{
++ if (try_warm && soc_ac97_ops.warm_reset) {
++ soc_ac97_ops.warm_reset(codec->ac97);
++ if (!(ac97_read(codec, 0) & 0x8000))
++ return 1;
++ }
++
++ soc_ac97_ops.reset(codec->ac97);
++ if (ac97_read(codec, 0) & 0x8000)
++ return -EIO;
++ return 0;
++}
++EXPORT_SYMBOL_GPL(wm9713_reset);
++
++static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
++{
++ u16 reg;
++
++ switch (event) {
++ case SNDRV_CTL_POWER_D0: /* full On */
++ /* enable thermal shutdown */
++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
++ ac97_write(codec, AC97_EXTENDED_MID, reg);
++ break;
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* enable master bias and vmid */
++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
++ ac97_write(codec, AC97_EXTENDED_MID, reg);
++ ac97_write(codec, AC97_POWERDOWN, 0x0000);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* disable everything including AC link */
++ ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
++ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
++ ac97_write(codec, AC97_POWERDOWN, 0xffff);
++ break;
++ }
++ codec->dapm_state = event;
++ return 0;
++}
++
++static int wm9713_soc_suspend(struct platform_device *pdev,
++ pm_message_t state)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ struct wm9713 *wm = (struct wm9713*)codec->private_data;
++
++ if (wm->pll) {
++ wm->pll_resume = wm->pll;
++ wm9713_set_pll(codec, 0);
++ }
++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++ return 0;
++}
++
++static int wm9713_soc_resume(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++ struct wm9713 *wm = (struct wm9713*)codec->private_data;
++ int i, ret;
++ u16 *cache = codec->reg_cache;
++
++ if ((ret = wm9713_reset(codec, 1)) < 0){
++ printk(KERN_ERR "could not reset AC97 codec\n");
++ return ret;
++ }
++
++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* only synchronise the codec if warm reset failed */
++ if (ret == 0) {
++ for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i+=2) {
++ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID ||
++ i == AC97_EXTENDED_MSTATUS || i > 0x66)
++ continue;
++ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
++ }
++ }
++
++ if (wm->pll_resume) {
++ wm9713_set_pll(codec, wm->pll_resume);
++ wm->pll_resume = 0;
++ }
++
++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
++
++ return ret;
++}
++
++static int wm9713_soc_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec;
++ int ret = 0, reg;
++
++ printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
++
++ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (socdev->codec == NULL)
++ return -ENOMEM;
++ codec = socdev->codec;
++ mutex_init(&codec->mutex);
++
++ codec->reg_cache =
++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9713_reg), GFP_KERNEL);
++ if (codec->reg_cache == NULL){
++ kfree(socdev->codec);
++ socdev->codec = NULL;
++ return -ENOMEM;
++ }
++ memcpy(codec->reg_cache, wm9713_reg,
++ sizeof(u16) * ARRAY_SIZE(wm9713_reg));
++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9713_reg);
++ codec->reg_cache_step = 2;
++
++ codec->private_data = kzalloc(sizeof(struct wm9713), GFP_KERNEL);
++ if (codec->private_data == NULL) {
++ kfree(codec->reg_cache);
++ kfree(socdev->codec);
++ socdev->codec = NULL;
++ return -ENOMEM;
++ }
++
++ codec->name = "WM9713";
++ codec->owner = THIS_MODULE;
++ codec->dai = wm9713_dai;
++ codec->num_dai = ARRAY_SIZE(wm9713_dai);
++ codec->write = ac97_write;
++ codec->read = ac97_read;
++ codec->dapm_event = wm9713_dapm_event;
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++
++ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
++ if (ret < 0)
++ goto err;
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0)
++ goto pcm_err;
++
++ /* do a cold reset for the controller and then try
++ * a warm reset followed by an optional cold reset for codec */
++ wm9713_reset(codec, 0);
++ ret = wm9713_reset(codec, 1);
++ if (ret < 0) {
++ printk(KERN_ERR "AC97 link error\n");
++ goto reset_err;
++ }
++
++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++ /* unmute the adc - move to kcontrol */
++ reg = ac97_read(codec, AC97_CD) & 0x7fff;
++ ac97_write(codec, AC97_CD, reg);
++
++ wm9713_add_controls(codec);
++ wm9713_add_widgets(codec);
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0)
++ goto reset_err;
++ return 0;
++
++reset_err:
++ snd_soc_free_pcms(socdev);
++
++pcm_err:
++ snd_soc_free_ac97_codec(codec);
++
++err:
++ kfree(socdev->codec->private_data);
++ kfree(socdev->codec->reg_cache);
++ kfree(socdev->codec);
++ socdev->codec = NULL;
++ return ret;
++}
++
++static int wm9713_soc_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (codec == NULL)
++ return 0;
++
++ snd_soc_dapm_free(socdev);
++ snd_soc_free_pcms(socdev);
++ snd_soc_free_ac97_codec(codec);
++ kfree(codec->private_data);
++ kfree(codec->reg_cache);
++ kfree(codec);
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm9713= {
++ .probe = wm9713_soc_probe,
++ .remove = wm9713_soc_remove,
++ .suspend = wm9713_soc_suspend,
++ .resume = wm9713_soc_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713);
++
++MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/codecs/wm9713.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/codecs/wm9713.h
+@@ -0,0 +1,18 @@
++/*
++ * wm9713.h -- WM9713 Soc Audio driver
++ */
++
++#ifndef _WM9713_H
++#define _WM9713_H
++
++#define WM9713_DAI_AC97_HIFI 0
++#define WM9713_DAI_AC97_AUX 1
++#define WM9713_DAI_PCM_VOICE 2
++
++extern struct snd_soc_codec_device soc_codec_dev_wm9713;
++extern struct snd_soc_codec_dai wm9713_dai[3];
++
++u32 wm9713_set_pll(struct snd_soc_codec *codec, u32 in);
++int wm9713_reset(struct snd_soc_codec *codec, int try_warm);
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/pxa/Kconfig
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/Kconfig
+@@ -0,0 +1,125 @@
++menu "SoC Audio for the Intel PXA2xx"
++
++config SND_PXA2xx_SOC
++ tristate "SoC Audio for the Intel PXA2xx chip"
++ depends on ARCH_PXA && SND
++ select SND_PCM
++ help
++ Say Y or M if you want to add support for codecs attached to
++ the PXA2xx AC97, I2S or SSP interface. You will also need
++ to select the audio interfaces to support below.
++
++config SND_PXA2xx_AC97
++ tristate
++ select SND_AC97_CODEC
++
++config SND_PXA2xx_SOC_AC97
++ tristate
++ select SND_AC97_BUS
++ select SND_SOC_AC97_BUS
++
++config SND_PXA2xx_SOC_I2S
++ tristate
++
++config SND_PXA2xx_SOC_SSP
++ tristate
++ select PXA_SSP
++
++config SND_PXA2xx_SOC_MAINSTONE
++ tristate "SoC AC97 Audio support for Intel Mainstone"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_AC97
++ help
++ Say Y if you want to add support for generic AC97 SoC audio on Mainstone.
++
++config SND_PXA2xx_SOC_MAINSTONE_WM8731
++ tristate "SoC I2S Audio support for Intel Mainstone - WM8731"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_I2S
++ help
++ Say Y if you want to add support for SoC audio on Mainstone
++ with the WM8731.
++
++config SND_PXA2xx_SOC_MAINSTONE_WM8753
++ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM8753"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_I2S
++ select SND_PXA2xx_SOC_SSP
++ help
++ Say Y if you want to add support for SoC audio on Mainstone
++ with the WM8753.
++
++config SND_PXA2xx_SOC_MAINSTONE_WM8974
++ tristate "SoC I2S Audio support for Intel Mainstone - WM8974"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_I2S
++ help
++ Say Y if you want to add support for SoC audio on Mainstone
++ with the WM8974.
++
++config SND_PXA2xx_SOC_MAINSTONE_WM9713
++ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM9713"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_AC97
++ select SND_PXA2xx_SOC_SSP
++ help
++ Say Y if you want to add support for SoC voice audio on Mainstone
++ with the WM9713.
++
++config SND_MAINSTONE_BASEBAND
++ tristate "Example SoC Baseband Audio support for Intel Mainstone"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_AC97
++ help
++ Say Y if you want to add support for SoC baseband on Mainstone
++ with the WM9713 and example Baseband modem.
++
++config SND_MAINSTONE_BLUETOOTH
++ tristate "Example SoC Bluetooth Audio support for Intel Mainstone"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_I2S
++ help
++ Say Y if you want to add support for SoC bluetooth on Mainstone
++ with the WM8753 and example Bluetooth codec.
++
++config SND_PXA2xx_SOC_MAINSTONE_WM9712
++ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM9712"
++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE
++ select SND_PXA2xx_SOC_AC97
++ help
++ Say Y if you want to add support for SoC voice audio on Mainstone
++ with the WM9712.
++
++config SND_PXA2xx_SOC_CORGI
++ tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
++ depends on SND_PXA2xx_SOC && PXA_SHARP_C7xx
++ select SND_PXA2xx_SOC_I2S
++ help
++ Say Y if you want to add support for SoC audio on Sharp
++ Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
++
++config SND_PXA2xx_SOC_SPITZ
++ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
++ depends on SND_PXA2xx_SOC && PXA_SHARP_Cxx00
++ select SND_PXA2xx_SOC_I2S
++ help
++ Say Y if you want to add support for SoC audio on Sharp
++ Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
++
++config SND_PXA2xx_SOC_POODLE
++ tristate "SoC Audio support for Poodle"
++ depends on SND_PXA2xx_SOC && MACH_POODLE
++ select SND_PXA2xx_SOC_I2S
++ help
++ Say Y if you want to add support for SoC audio on Sharp
++ Zaurus SL-5600 model (Poodle).
++
++config SND_PXA2xx_SOC_TOSA
++ tristate "SoC AC97 Audio support for Tosa"
++ depends on SND_PXA2xx_SOC && MACH_TOSA
++ select SND_PXA2xx_SOC_AC97
++ help
++ Say Y if you want to add support for SoC audio on Sharp
++ Zaurus SL-C6000x models (Tosa).
++
++endmenu
+Index: linux-2.6-pxa-new/sound/soc/pxa/Makefile
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/Makefile
+@@ -0,0 +1,36 @@
++# PXA Platform Support
++snd-soc-pxa2xx-objs := pxa2xx-pcm.o
++snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
++snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
++snd-soc-pxa2xx-ssp-objs := pxa2xx-ssp.o
++
++obj-$(CONFIG_SND_PXA2xx_SOC) += snd-soc-pxa2xx.o
++obj-$(CONFIG_SND_PXA2xx_SOC_AC97) += snd-soc-pxa2xx-ac97.o
++obj-$(CONFIG_SND_PXA2xx_SOC_I2S) += snd-soc-pxa2xx-i2s.o
++obj-$(CONFIG_SND_PXA2xx_SOC_SSP) += snd-soc-pxa2xx-ssp.o
++
++# PXA Machine Support
++snd-soc-corgi-objs := corgi.o
++snd-soc-mainstone-wm8731-objs := mainstone_wm8731.o
++snd-soc-mainstone-wm8753-objs := mainstone_wm8753.o
++snd-soc-mainstone-wm8974-objs := mainstone_wm8974.o
++snd-soc-mainstone-wm9713-objs := mainstone_wm9713.o
++snd-soc-mainstone-wm9712-objs := mainstone_wm9712.o
++snd-soc-mainstone-baseband-objs := mainstone_baseband.o
++snd-soc-mainstone-bluetooth-objs := mainstone_bluetooth.o
++snd-soc-poodle-objs := poodle.o
++snd-soc-tosa-objs := tosa.o
++snd-soc-spitz-objs := spitz.o
++
++obj-$(CONFIG_SND_PXA2xx_SOC_CORGI) += snd-soc-corgi.o
++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8731) += snd-soc-mainstone-wm8731.o
++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8753) += snd-soc-mainstone-wm8753.o
++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8974) += snd-soc-mainstone-wm8974.o
++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM9713) += snd-soc-mainstone-wm9713.o
++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM9712) += snd-soc-mainstone-wm9712.o
++obj-$(CONFIG_SND_MAINSTONE_BASEBAND) += snd-soc-mainstone-baseband.o
++obj-$(CONFIG_SND_MAINSTONE_BLUETOOTH) += snd-soc-mainstone-bluetooth.o
++obj-$(CONFIG_SND_PXA2xx_SOC_POODLE) += snd-soc-poodle.o
++obj-$(CONFIG_SND_PXA2xx_SOC_TOSA) += snd-soc-tosa.o
++obj-$(CONFIG_SND_PXA2xx_SOC_SPITZ) += snd-soc-spitz.o
++
+Index: linux-2.6-pxa-new/sound/soc/pxa/corgi.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/corgi.c
+@@ -0,0 +1,361 @@
++/*
++ * corgi.c -- SoC audio for Corgi
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
++ * Richard Purdie <richard@openedhand.com>
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 30th Nov 2005 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/timer.h>
++#include <linux/interrupt.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/mach-types.h>
++#include <asm/hardware/scoop.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/hardware.h>
++#include <asm/arch/corgi.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm8731.h"
++#include "pxa2xx-pcm.h"
++
++#define CORGI_HP 0
++#define CORGI_MIC 1
++#define CORGI_LINE 2
++#define CORGI_HEADSET 3
++#define CORGI_HP_OFF 4
++#define CORGI_SPK_ON 0
++#define CORGI_SPK_OFF 1
++
++ /* audio clock in Hz - rounded from 12.235MHz */
++#define CORGI_AUDIO_CLOCK 12288000
++
++static int corgi_jack_func;
++static int corgi_spk_func;
++
++static void corgi_ext_control(struct snd_soc_codec *codec)
++{
++ int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
++
++ /* set up jack connection */
++ switch (corgi_jack_func) {
++ case CORGI_HP:
++ hp = 1;
++ /* set = unmute headphone */
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
++ break;
++ case CORGI_MIC:
++ mic = 1;
++ /* reset = mute headphone */
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
++ break;
++ case CORGI_LINE:
++ line = 1;
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
++ break;
++ case CORGI_HEADSET:
++ hs = 1;
++ mic = 1;
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
++ break;
++ }
++
++ if (corgi_spk_func == CORGI_SPK_ON)
++ spk = 1;
++
++ /* set the enpoints to their new connetion states */
++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic);
++ snd_soc_dapm_set_endpoint(codec, "Line Jack", line);
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
++
++ /* signal a DAPM event */
++ snd_soc_dapm_sync_endpoints(codec);
++}
++
++static int corgi_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec *codec = rtd->socdev->codec;
++
++ /* check the jack status at stream startup */
++ corgi_ext_control(codec);
++ return 0;
++}
++
++/* we need to unmute the HP at shutdown as the mute burns power on corgi */
++static int corgi_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec *codec = rtd->socdev->codec;
++
++ /* set = unmute headphone */
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
++ return 0;
++}
++
++static struct snd_soc_ops corgi_ops = {
++ .startup = corgi_startup,
++ .shutdown = corgi_shutdown,
++};
++
++static int corgi_get_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = corgi_jack_func;
++ return 0;
++}
++
++static int corgi_set_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (corgi_jack_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ corgi_jack_func = ucontrol->value.integer.value[0];
++ corgi_ext_control(codec);
++ return 1;
++}
++
++static int corgi_get_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = corgi_spk_func;
++ return 0;
++}
++
++static int corgi_set_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (corgi_spk_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ corgi_spk_func = ucontrol->value.integer.value[0];
++ corgi_ext_control(codec);
++ return 1;
++}
++
++static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
++{
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
++ else
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
++
++ return 0;
++}
++
++static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event)
++{
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
++ else
++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
++
++ return 0;
++}
++
++/* corgi machine dapm widgets */
++static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
++SND_SOC_DAPM_HP("Headphone Jack", NULL),
++SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
++SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
++SND_SOC_DAPM_LINE("Line Jack", NULL),
++SND_SOC_DAPM_HP("Headset Jack", NULL),
++};
++
++/* Corgi machine audio map (connections to the codec pins) */
++static const char *audio_map[][3] = {
++
++ /* headset Jack - in = micin, out = LHPOUT*/
++ {"Headset Jack", NULL, "LHPOUT"},
++
++ /* headphone connected to LHPOUT1, RHPOUT1 */
++ {"Headphone Jack", NULL, "LHPOUT"},
++ {"Headphone Jack", NULL, "RHPOUT"},
++
++ /* speaker connected to LOUT, ROUT */
++ {"Ext Spk", NULL, "ROUT"},
++ {"Ext Spk", NULL, "LOUT"},
++
++ /* mic is connected to MICIN (via right channel of headphone jack) */
++ {"MICIN", NULL, "Mic Jack"},
++
++ /* Same as the above but no mic bias for line signals */
++ {"MICIN", NULL, "Line Jack"},
++
++ {NULL, NULL, NULL},
++};
++
++static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
++ "Off"};
++static const char *spk_function[] = {"On", "Off"};
++static const struct soc_enum corgi_enum[] = {
++ SOC_ENUM_SINGLE_EXT(5, jack_function),
++ SOC_ENUM_SINGLE_EXT(2, spk_function),
++};
++
++static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
++ SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
++ corgi_set_jack),
++ SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
++ corgi_set_spk),
++};
++
++/*
++ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
++ */
++static int corgi_wm8731_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
++
++ /* Add corgi specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ /* Add corgi specific widgets */
++ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
++ }
++
++ /* Set up corgi specific audio path audio_map */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++static unsigned int corgi_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) {
++ /* pxa2xx is i2s master */
++ switch (info->rate) {
++ case 44100:
++ case 88200:
++ /* configure codec digital filters for 44.1, 88.2 */
++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ 11289600);
++ break;
++ default:
++ /* configure codec digital filters for all other rates */
++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ CORGI_AUDIO_CLOCK);
++ break;
++ }
++ /* config pxa i2s as master */
++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info,
++ CORGI_AUDIO_CLOCK);
++ } else {
++ /* codec is i2s master -
++ * only configure codec DAI clock and filters */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ CORGI_AUDIO_CLOCK);
++ }
++}
++
++/* corgi digital audio interface glue - connects codec <--> CPU */
++static struct snd_soc_dai_link corgi_dai = {
++ .name = "WM8731",
++ .stream_name = "WM8731",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8731_dai,
++ .init = corgi_wm8731_init,
++ .config_sysclk = corgi_config_sysclk,
++};
++
++/* corgi audio machine driver */
++static struct snd_soc_machine snd_soc_machine_corgi = {
++ .name = "Corgi",
++ .dai_link = &corgi_dai,
++ .num_links = 1,
++ .ops = &corgi_ops,
++};
++
++/* corgi audio private data */
++static struct wm8731_setup_data corgi_wm8731_setup = {
++ .i2c_address = 0x1b,
++};
++
++/* corgi audio subsystem */
++static struct snd_soc_device corgi_snd_devdata = {
++ .machine = &snd_soc_machine_corgi,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8731,
++ .codec_data = &corgi_wm8731_setup,
++};
++
++static struct platform_device *corgi_snd_device;
++
++static int __init corgi_init(void)
++{
++ int ret;
++
++ if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky()))
++ return -ENODEV;
++
++ corgi_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!corgi_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata);
++ corgi_snd_devdata.dev = &corgi_snd_device->dev;
++ ret = platform_device_add(corgi_snd_device);
++
++ if (ret)
++ platform_device_put(corgi_snd_device);
++
++ return ret;
++}
++
++static void __exit corgi_exit(void)
++{
++ platform_device_unregister(corgi_snd_device);
++}
++
++module_init(corgi_init);
++module_exit(corgi_exit);
++
++/* Module information */
++MODULE_AUTHOR("Richard Purdie");
++MODULE_DESCRIPTION("ALSA SoC Corgi");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone.c
+@@ -0,0 +1,126 @@
++/*
++ * mainstone.c -- SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
++ *
++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 30th Oct 2005 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/mainstone.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/ac97.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++static long mst_audio_suspend_mask;
++
++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ mst_audio_suspend_mask = MST_MSCWR2;
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_resume(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_probe(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_remove(struct platform_device *pdev)
++{
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static struct snd_soc_machine_config codecs[] = {
++{
++ .name = "AC97",
++ .sname = "AC97 HiFi",
++ .iface = &pxa_ac97_interface[0],
++},
++{
++ .name = "AC97 Aux",
++ .sname = "AC97 Aux",
++ .iface = &pxa_ac97_interface[1],
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .probe = mainstone_probe,
++ .remove = mainstone_remove,
++ .suspend_pre = mainstone_suspend,
++ .resume_post = mainstone_resume,
++ .config = codecs,
++ .nconfigs = ARRAY_SIZE(codecs),
++};
++
++static struct snd_soc_device mainstone_snd_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_ac97,
++};
++
++static struct platform_device *mainstone_snd_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
++ ret = platform_device_add(mainstone_snd_device);
++
++ if (ret)
++ platform_device_put(mainstone_snd_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_baseband.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_baseband.c
+@@ -0,0 +1,249 @@
++/*
++ * mainstone_baseband.c
++ * Mainstone Example Baseband modem -- ALSA Soc Audio Layer
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 15th Apr 2006 Initial version.
++ *
++ * This is example code to demonstrate connecting a baseband modem to the PCM
++ * DAI on the WM9713 codec on the Intel Mainstone platform. It is by no means
++ * complete as it requires code to control the modem.
++ *
++ * The architecture consists of the WM9713 AC97 DAI connected to the PXA27x
++ * AC97 controller and the WM9713 PCM DAI connected to the basebands DAI. The
++ * baseband is controlled via a serial port. Audio is routed between the PXA27x
++ * and the baseband via internal WM9713 analog paths.
++ *
++ * This driver is not the baseband modem driver. This driver only calls
++ * functions from the Baseband driver to set up it's PCM DAI.
++ *
++ * It's intended to use this driver as follows:-
++ *
++ * 1. open() WM9713 PCM audio device.
++ * 2. open() serial device (for AT commands).
++ * 3. configure PCM audio device (rate etc) - sets up WM9713 PCM DAI,
++ * this will also set up the baseband PCM DAI (via calling baseband driver).
++ * 4. send any further AT commands to set up baseband.
++ * 5. configure codec audio mixer paths.
++ * 6. open(), configure and read/write AC97 audio device - to Tx/Rx voice
++ *
++ * The PCM audio device is opened but IO is never performed on it as the IO is
++ * directly between the codec and the baseband (and not the CPU).
++ *
++ * TODO:
++ * o Implement callbacks
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++#include <asm/arch/ssp.h>
++
++#include "../codecs/wm9713.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++
++#define BASEBAND_XXX_DAIFMT \
++ (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\
++ SND_SOC_DAIFMT_NB_NF)
++
++#define BASEBAND_XXX_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++/*
++ * PCM modes - 8k 16bit mono baseband modem is master
++ */
++static struct snd_soc_dai_mode mainstone_example_modes[] = {
++ /* port master clk & frame modes */
++ {BASEBAND_XXX_DAIFMT, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
++ SNDRV_PCM_RATE_8000, BASEBAND_XXX_DIR, SND_SOC_DAI_BFS_RATE, 256, 64},
++};
++
++/* Do specific baseband PCM voice startup here */
++static int mainstone_baseband_startup(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++/* Do specific baseband PCM voice shutdown here */
++static void mainstone_baseband_shutdown (struct snd_pcm_substream *substream)
++{
++}
++
++/* Do specific baseband modem PCM voice hw params init here */
++static int mainstone_baseband_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ return 0;
++}
++
++/* Do specific baseband modem PCM voice hw params free here */
++static int mainstone_baseband_hw_free(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static struct snd_soc_cpu_dai mainstone_example_dai[] = {
++ { .name = "Baseband",
++ .id = 0,
++ .type = SND_SOC_DAI_PCM,
++ .playback = {
++ .channels_min = 1,
++ .channels_max = 1,},
++ .capture = {
++ .channels_min = 1,
++ .channels_max = 1,},
++ .ops = {
++ .startup = mainstone_baseband_startup,
++ .shutdown = mainstone_baseband_shutdown,
++ .hw_params = mainstone_baseband_hw_params,
++ .hw_free = mainstone_baseband_hw_free,
++ },
++ .caps = {
++ .mode = mainstone_example_modes,
++ .num_modes = ARRAY_SIZE(mainstone_example_modes),},
++ },
++};
++
++/* do we need to do any thing on the mainstone when the stream is
++ * started and stopped
++ */
++static int mainstone_startup(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static void mainstone_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++static struct snd_soc_ops mainstone_ops = {
++ .startup = mainstone_startup,
++ .shutdown = mainstone_shutdown,
++};
++
++/* PM */
++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ return 0;
++}
++
++static int mainstone_resume(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mainstone_probe(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mainstone_remove(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mainstone_wm9713_init(struct snd_soc_codec *codec)
++{
++ return 0;
++}
++
++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ /* wm8753 has pll that generates mclk from 13MHz xtal */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000);
++}
++
++/* the physical audio connections between the WM9713, Baseband and pxa2xx */
++static struct snd_soc_dai_link mainstone_dai[] = {
++{
++ .name = "AC97",
++ .stream_name = "AC97 HiFi",
++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
++ .init = mainstone_wm9713_init,
++},
++{
++ .name = "AC97 Aux",
++ .stream_name = "AC97 Aux",
++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
++},
++{
++ .name = "Baseband",
++ .stream_name = "Voice",
++ .cpu_dai = mainstone_example_dai,
++ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
++ .config_sysclk = mainstone_config_sysclk,
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .probe = mainstone_probe,
++ .remove = mainstone_remove,
++ .suspend_pre = mainstone_suspend,
++ .resume_post = mainstone_resume,
++ .ops = &mainstone_ops,
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct snd_soc_device mainstone_snd_ac97_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm9713,
++};
++
++static struct platform_device *mainstone_snd_ac97_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_ac97_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata);
++ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev;
++
++ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0)
++ platform_device_put(mainstone_snd_ac97_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_ac97_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("Mainstone Example Baseband PCM Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_bluetooth.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_bluetooth.c
+@@ -0,0 +1,399 @@
++/*
++ * mainstone_bluetooth.c
++ * Mainstone Example Bluetooth -- ALSA Soc Audio Layer
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 15th May 2006 Initial version.
++ *
++ * This is example code to demonstrate connecting a bluetooth codec to the PCM
++ * DAI on the WM8753 codec on the Intel Mainstone platform. It is by no means
++ * complete as it requires code to control the BT codec.
++ *
++ * The architecture consists of the WM8753 HIFI DAI connected to the PXA27x
++ * I2S controller and the WM8753 PCM DAI connected to the bluetooth DAI. The
++ * bluetooth codec and wm8753 are controlled via I2C. Audio is routed between
++ * the PXA27x and the bluetooth via internal WM8753 analog paths.
++ *
++ * This example supports the following audio input/outputs.
++ *
++ * o Board mounted Mic and Speaker (spk has amplifier)
++ * o Headphones via jack socket
++ * o BT source and sink
++ *
++ * This driver is not the bluetooth codec driver. This driver only calls
++ * functions from the Bluetooth driver to set up it's PCM DAI.
++ *
++ * It's intended to use the driver as follows:-
++ *
++ * 1. open() WM8753 PCM audio device.
++ * 2. configure PCM audio device (rate etc) - sets up WM8753 PCM DAI,
++ * this should also set up the BT codec DAI (via calling bt driver).
++ * 3. configure codec audio mixer paths.
++ * 4. open(), configure and read/write HIFI audio device - to Tx/Rx voice
++ *
++ * The PCM audio device is opened but IO is never performed on it as the IO is
++ * directly between the codec and the BT codec (and not the CPU).
++ *
++ * TODO:
++ * o Implement callbacks
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++#include <asm/arch/ssp.h>
++
++#include "../codecs/wm8753.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++
++#define BLUETOOTH_DAIFMT \
++ (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\
++ SND_SOC_DAIFMT_NB_NF)
++
++#define BLUETOOTH_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++/*
++ * PCM modes - 8k 16bit mono BT codec is master
++ */
++static struct snd_soc_dai_mode mainstone_bt_modes[] = {
++ /* port master clk & frame modes */
++ {BLUETOOTH_DAIFMT, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
++ SNDRV_PCM_RATE_8000, BLUETOOTH_DIR, SND_SOC_DAI_BFS_RATE, 256, 64},
++};
++
++/* Do specific bluetooth PCM startup here */
++static int mainstone_bt_startup(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++/* Do specific bluetooth PCM shutdown here */
++static void mainstone_bt_shutdown (struct snd_pcm_substream *substream)
++{
++}
++
++/* Do pecific bluetooth PCM hw params init here */
++static int mainstone_bt_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ return 0;
++}
++
++/* Do specific bluetooth PCM hw params free here */
++static int mainstone_bt_hw_free(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static struct snd_soc_cpu_dai mainstone_bt_dai[] = {
++ { .name = "Bluetooth",
++ .id = 0,
++ .type = SND_SOC_DAI_PCM,
++ .playback = {
++ .channels_min = 1,
++ .channels_max = 1,},
++ .capture = {
++ .channels_min = 1,
++ .channels_max = 1,},
++ .ops = {
++ .startup = mainstone_bt_startup,
++ .shutdown = mainstone_bt_shutdown,
++ .hw_params = mainstone_bt_hw_params,
++ .hw_free = mainstone_bt_hw_free,
++ },
++ .caps = {
++ .mode = mainstone_bt_modes,
++ .num_modes = ARRAY_SIZE(mainstone_bt_modes),},
++ },
++};
++
++/* do we need to do any thing on the mainstone when the stream is
++ * started and stopped
++ */
++static int mainstone_startup(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static void mainstone_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++static struct snd_soc_ops mainstone_ops = {
++ .startup = mainstone_startup,
++ .shutdown = mainstone_shutdown,
++};
++
++/* PM */
++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ return 0;
++}
++
++static int mainstone_resume(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mainstone_probe(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mainstone_remove(struct platform_device *pdev)
++{
++ return 0;
++}
++
++/*
++ * Machine audio functions.
++ *
++ * The machine now has 3 extra audio controls.
++ *
++ * Jack function: Sets function (device plugged into Jack) to nothing (Off)
++ * or Headphones.
++ *
++ * Mic function: Set the on board Mic to On or Off
++ * Spk function: Set the on board Spk to On or Off
++ *
++ * example: BT playback (of far end) and capture (of near end)
++ * Set Mic and Speaker to On, open BT alsa interface as above and set up
++ * internal audio paths.
++ */
++
++static int machine_jack_func = 0;
++static int machine_spk_func = 0;
++static int machine_mic_func = 0;
++
++static int machine_get_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = machine_jack_func;
++ return 0;
++}
++
++static int machine_set_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ machine_jack_func = ucontrol->value.integer.value[0];
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", machine_jack_func);
++ return 0;
++}
++
++static int machine_get_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = machine_spk_func;
++ return 0;
++}
++
++static int machine_set_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ machine_spk_func = ucontrol->value.integer.value[0];
++ snd_soc_dapm_set_endpoint(codec, "Spk", machine_spk_func);
++ return 0;
++}
++
++static int machine_get_mic(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = machine_spk_func;
++ return 0;
++}
++
++static int machine_set_mic(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ machine_spk_func = ucontrol->value.integer.value[0];
++ snd_soc_dapm_set_endpoint(codec, "Mic", machine_mic_func);
++ return 0;
++}
++
++/* turns on board speaker amp on/off */
++static int machine_amp_event(struct snd_soc_dapm_widget *w, int event)
++{
++#if 0
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ /* on */
++ else
++ /* off */
++#endif
++ return 0;
++}
++
++/* machine dapm widgets */
++static const struct snd_soc_dapm_widget machine_dapm_widgets[] = {
++SND_SOC_DAPM_HP("Headphone Jack", NULL),
++SND_SOC_DAPM_SPK("Spk", machine_amp_event),
++SND_SOC_DAPM_MIC("Mic", NULL),
++};
++
++/* machine connections to the codec pins */
++static const char* audio_map[][3] = {
++
++ /* headphone connected to LOUT1, ROUT1 */
++ {"Headphone Jack", NULL, "LOUT"},
++ {"Headphone Jack", NULL, "ROUT"},
++
++ /* speaker connected to LOUT2, ROUT2 */
++ {"Spk", NULL, "ROUT2"},
++ {"Spk", NULL, "LOUT2"},
++
++ /* mic is connected to MIC1 (via Mic Bias) */
++ {"MIC1", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic"},
++
++ {NULL, NULL, NULL},
++};
++
++static const char* jack_function[] = {"Off", "Headphone"};
++static const char* spk_function[] = {"Off", "On"};
++static const char* mic_function[] = {"Off", "On"};
++static const struct soc_enum machine_ctl_enum[] = {
++ SOC_ENUM_SINGLE_EXT(2, jack_function),
++ SOC_ENUM_SINGLE_EXT(2, spk_function),
++ SOC_ENUM_SINGLE_EXT(2, mic_function),
++};
++
++static const struct snd_kcontrol_new wm8753_machine_controls[] = {
++ SOC_ENUM_EXT("Jack Function", machine_ctl_enum[0], machine_get_jack, machine_set_jack),
++ SOC_ENUM_EXT("Speaker Function", machine_ctl_enum[1], machine_get_spk, machine_set_spk),
++ SOC_ENUM_EXT("Mic Function", machine_ctl_enum[2], machine_get_mic, machine_set_mic),
++};
++
++static int mainstone_wm8753_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ /* not used on this machine - e.g. will never be powered up */
++ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
++ snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
++ snd_soc_dapm_set_endpoint(codec, "MONO2", 0);
++ snd_soc_dapm_set_endpoint(codec, "MONO1", 0);
++ snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
++ snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
++ snd_soc_dapm_set_endpoint(codec, "RXP", 0);
++ snd_soc_dapm_set_endpoint(codec, "RXN", 0);
++ snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
++
++ /* Add machine specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8753_machine_controls); i++) {
++ if ((err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8753_machine_controls[i],codec, NULL))) < 0)
++ return err;
++ }
++
++ /* Add machine specific widgets */
++ for(i = 0; i < ARRAY_SIZE(machine_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &machine_dapm_widgets[i]);
++ }
++
++ /* Set up machine specific audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++/* this configures the clocking between the WM8753 and the BT codec */
++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ /* wm8753 has pll that generates mclk from 13MHz xtal */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000);
++}
++
++static struct snd_soc_dai_link mainstone_dai[] = {
++{ /* Hifi Playback - for similatious use with voice below */
++ .name = "WM8753",
++ .stream_name = "WM8753 HiFi",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
++ .init = mainstone_wm8753_init,
++ .config_sysclk = mainstone_config_sysclk,
++},
++{ /* Voice via BT */
++ .name = "Bluetooth",
++ .stream_name = "Voice",
++ .cpu_dai = mainstone_bt_dai,
++ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
++ .config_sysclk = mainstone_config_sysclk,
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .probe = mainstone_probe,
++ .remove = mainstone_remove,
++ .suspend_pre = mainstone_suspend,
++ .resume_post = mainstone_resume,
++ .ops = &mainstone_ops,
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct snd_soc_device mainstone_snd_wm8753_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8753,
++};
++
++static struct platform_device *mainstone_snd_wm8753_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_wm8753_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_wm8753_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_wm8753_device, &mainstone_snd_wm8753_devdata);
++ mainstone_snd_wm8753_devdata.dev = &mainstone_snd_wm8753_device->dev;
++
++ if((ret = platform_device_add(mainstone_snd_wm8753_device)) != 0)
++ platform_device_put(mainstone_snd_wm8753_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_wm8753_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("Mainstone Example Bluetooth PCM Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8731.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8731.c
+@@ -0,0 +1,156 @@
++/*
++ * mainstone.c -- SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
++ *
++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 5th June 2006 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/mainstone.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm8731.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++
++
++static const struct snd_soc_dapm_widget dapm_widgets[] = {
++ SND_SOC_DAPM_MIC("Int Mic", NULL),
++ SND_SOC_DAPM_SPK("Ext Spk", NULL),
++};
++
++static const char* intercon[][3] = {
++
++ /* speaker connected to LHPOUT */
++ {"Ext Spk", NULL, "LHPOUT"},
++
++ /* mic is connected to Mic Jack, with WM8731 Mic Bias */
++ {"MICIN", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Int Mic"},
++
++ /* terminator */
++ {NULL, NULL, NULL},
++};
++
++/*
++ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
++ */
++static int mainstone_wm8731_init(struct snd_soc_codec *codec)
++{
++ int i;
++
++
++ /* Add specific widgets */
++ for(i = 0; i < ARRAY_SIZE(dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &dapm_widgets[i]);
++ }
++
++ /* Set up specific audio path interconnects */
++ for(i = 0; intercon[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]);
++ }
++
++ /* not connected */
++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
++
++ /* always connected */
++ snd_soc_dapm_set_endpoint(codec, "Int Mic", 1);
++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
++
++ snd_soc_dapm_sync_endpoints(codec);
++
++ return 0;
++}
++
++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ /* we have a 12.288MHz crystal */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12288000);
++}
++
++static struct snd_soc_dai_link mainstone_dai[] = {
++{
++ .name = "WM8731",
++ .stream_name = "WM8731 HiFi",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8731_dai,
++ .init = mainstone_wm8731_init,
++ .config_sysclk = mainstone_config_sysclk,
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct wm8731_setup_data corgi_wm8731_setup = {
++ .i2c_address = 0x1b,
++};
++
++static struct snd_soc_device mainstone_snd_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8731,
++ .codec_data = &corgi_wm8731_setup,
++};
++
++static struct platform_device *mainstone_snd_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
++ ret = platform_device_add(mainstone_snd_device);
++
++ if (ret)
++ platform_device_put(mainstone_snd_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM8731 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8753.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8753.c
+@@ -0,0 +1,226 @@
++/*
++ * mainstone.c -- SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
++ *
++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 30th Oct 2005 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/mainstone.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm8753.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) {
++ /* enable USB on the go MUX so we can use SSPFRM2 */
++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL;
++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST;
++ }
++ return 0;
++}
++
++static void mainstone_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) {
++ /* disable USB on the go MUX so we can use ttyS0 */
++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL;
++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST;
++ }
++}
++
++static struct snd_soc_ops mainstone_ops = {
++ .startup = mainstone_startup,
++ .shutdown = mainstone_shutdown,
++};
++
++static long mst_audio_suspend_mask;
++
++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ mst_audio_suspend_mask = MST_MSCWR2;
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_resume(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_probe(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_remove(struct platform_device *pdev)
++{
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++/* example machine audio_mapnections */
++static const char* audio_map[][3] = {
++
++ /* mic is connected to mic1 - with bias */
++ {"MIC1", NULL, "Mic Bias"},
++ {"MIC1N", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic1 Jack"},
++ {"Mic Bias", NULL, "Mic1 Jack"},
++
++ {"ACIN", NULL, "ACOP"},
++ {NULL, NULL, NULL},
++};
++
++/* headphone detect support on my board */
++static const char * hp_pol[] = {"Headphone", "Speaker"};
++static const struct soc_enum wm8753_enum =
++ SOC_ENUM_SINGLE(WM8753_OUTCTL, 1, 2, hp_pol);
++
++static const struct snd_kcontrol_new wm8753_mainstone_controls[] = {
++ SOC_SINGLE("Headphone Detect Switch", WM8753_OUTCTL, 6, 1, 0),
++ SOC_ENUM("Headphone Detect Polarity", wm8753_enum),
++};
++
++/*
++ * This is an example machine initialisation for a wm8753 connected to a
++ * Mainstone II. It is missing logic to detect hp/mic insertions and logic
++ * to re-route the audio in such an event.
++ */
++static int mainstone_wm8753_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ /* set up mainstone codec pins */
++ snd_soc_dapm_set_endpoint(codec, "RXP", 0);
++ snd_soc_dapm_set_endpoint(codec, "RXN", 0);
++ snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
++
++ /* add mainstone specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8753_mainstone_controls); i++) {
++ if ((err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8753_mainstone_controls[i],codec, NULL))) < 0)
++ return err;
++ }
++
++ /* set up mainstone specific audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ /* wm8753 has pll that generates mclk from 13MHz xtal */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000);
++}
++
++static struct snd_soc_dai_link mainstone_dai[] = {
++{ /* Hifi Playback - for similatious use with voice below */
++ .name = "WM8753",
++ .stream_name = "WM8753 HiFi",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
++ .init = mainstone_wm8753_init,
++ .config_sysclk = mainstone_config_sysclk,
++},
++{ /* Voice via BT */
++ .name = "Bluetooth",
++ .stream_name = "Voice",
++ .cpu_dai = &pxa_ssp_dai[1],
++ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
++ .config_sysclk = mainstone_config_sysclk,
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .probe = mainstone_probe,
++ .remove = mainstone_remove,
++ .suspend_pre = mainstone_suspend,
++ .resume_post = mainstone_resume,
++ .ops = &mainstone_ops,
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct wm8753_setup_data mainstone_wm8753_setup = {
++ .i2c_address = 0x1a,
++};
++
++static struct snd_soc_device mainstone_snd_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8753,
++ .codec_data = &mainstone_wm8753_setup,
++};
++
++static struct platform_device *mainstone_snd_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
++ ret = platform_device_add(mainstone_snd_device);
++
++ if (ret)
++ platform_device_put(mainstone_snd_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM8753 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8974.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8974.c
+@@ -0,0 +1,112 @@
++/*
++ * mainstone.c -- SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
++ *
++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 30th Oct 2005 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/mainstone.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm8974.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_wm8974_init(struct snd_soc_codec *codec)
++{
++ return 0;
++}
++
++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ /* we have a PLL */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12288000);
++
++}
++
++static struct snd_soc_dai_link mainstone_dai[] = {
++{
++ .name = "WM8974",
++ .stream_name = "WM8974 HiFi",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8974_dai,
++ .init = mainstone_wm8974_init,
++ .config_sysclk = mainstone_config_sysclk,
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct wm8974_setup_data mainstone_wm8974_setup = {
++ .i2c_address = 0x1a,
++};
++
++static struct snd_soc_device mainstone_snd_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8974,
++ .codec_data = &mainstone_wm8974_setup,
++};
++
++static struct platform_device *mainstone_snd_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
++ ret = platform_device_add(mainstone_snd_device);
++
++ if (ret)
++ platform_device_put(mainstone_snd_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9712.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9712.c
+@@ -0,0 +1,171 @@
++/*
++ * mainstone.c -- SoC audio for Mainstone
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
++ *
++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 29th Jan 2006 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/mainstone.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm9712.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++static long mst_audio_suspend_mask;
++
++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ mst_audio_suspend_mask = MST_MSCWR2;
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_resume(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_probe(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_remove(struct platform_device *pdev)
++{
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++/* mainstone machine dapm widgets */
++static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = {
++ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
++};
++
++/* example machine interconnections */
++static const char* intercon[][3] = {
++
++ /* mic is connected to mic1 - with bias */
++ {"MIC1", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic (Internal)"},
++
++ {NULL, NULL, NULL},
++};
++
++/*
++ * This is an example machine initialisation for a wm8753 connected to a
++ * Mainstone II. It is missing logic to detect hp/mic insertions and logic
++ * to re-route the audio in such an event.
++ */
++static int mainstone_wm9712_init(struct snd_soc_codec *codec)
++{
++ int i;
++
++ /* set up mainstone codec pins */
++ snd_soc_dapm_set_endpoint(codec, "RXP", 0);
++ snd_soc_dapm_set_endpoint(codec, "RXN", 0);
++ //snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
++
++ /* Add mainstone specific widgets */
++ for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]);
++ }
++
++ /* set up mainstone specific audio path interconnects */
++ for(i = 0; intercon[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++static struct snd_soc_dai_link mainstone_dai[] = {
++{
++ .name = "AC97",
++ .stream_name = "AC97 HiFi",
++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
++ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
++ .init = mainstone_wm9712_init,
++},
++{
++ .name = "AC97 Aux",
++ .stream_name = "AC97 Aux",
++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
++ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .probe = mainstone_probe,
++ .remove = mainstone_remove,
++ .suspend_pre = mainstone_suspend,
++ .resume_post = mainstone_resume,
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct snd_soc_device mainstone_snd_ac97_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm9712,
++};
++
++static struct platform_device *mainstone_snd_ac97_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_ac97_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata);
++ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev;
++
++ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0)
++ platform_device_put(mainstone_snd_ac97_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_ac97_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM9712 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9713.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9713.c
+@@ -0,0 +1,263 @@
++/*
++ * mainstone.c -- SoC audio for Mainstone
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
++ *
++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 29th Jan 2006 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/mainstone.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm9713.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) {
++ /* enable USB on the go MUX so we can use SSPFRM2 */
++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL;
++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST;
++ }
++ return 0;
++}
++
++static void mainstone_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) {
++ /* disable USB on the go MUX so we can use ttyS0 */
++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL;
++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST;
++ }
++}
++
++static struct snd_soc_ops mainstone_ops = {
++ .startup = mainstone_startup,
++ .shutdown = mainstone_shutdown,
++};
++
++static int test = 0;
++static int get_test(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = test;
++ return 0;
++}
++
++static int set_test(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ test = ucontrol->value.integer.value[0];
++ if(test) {
++
++ } else {
++
++ }
++ return 0;
++}
++
++static long mst_audio_suspend_mask;
++
++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ mst_audio_suspend_mask = MST_MSCWR2;
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_resume(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_probe(struct platform_device *pdev)
++{
++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static int mainstone_remove(struct platform_device *pdev)
++{
++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
++ return 0;
++}
++
++static const char* test_function[] = {"Off", "On"};
++static const struct soc_enum mainstone_enum[] = {
++ SOC_ENUM_SINGLE_EXT(2, test_function),
++};
++
++static const struct snd_kcontrol_new mainstone_controls[] = {
++ SOC_ENUM_EXT("ATest Function", mainstone_enum[0], get_test, set_test),
++};
++
++/* mainstone machine dapm widgets */
++static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = {
++ SND_SOC_DAPM_MIC("Mic 1", NULL),
++ SND_SOC_DAPM_MIC("Mic 2", NULL),
++ SND_SOC_DAPM_MIC("Mic 3", NULL),
++};
++
++/* example machine audio_mapnections */
++static const char* audio_map[][3] = {
++
++ /* mic is connected to mic1 - with bias */
++ {"MIC1", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic 1"},
++ /* mic is connected to mic2A - with bias */
++ {"MIC2A", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic 2"},
++ /* mic is connected to mic2B - with bias */
++ {"MIC2B", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic 3"},
++
++ {NULL, NULL, NULL},
++};
++
++/*
++ * This is an example machine initialisation for a wm9713 connected to a
++ * Mainstone II. It is missing logic to detect hp/mic insertions and logic
++ * to re-route the audio in such an event.
++ */
++static int mainstone_wm9713_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ /* set up mainstone codec pins */
++ snd_soc_dapm_set_endpoint(codec, "RXP", 0);
++ snd_soc_dapm_set_endpoint(codec, "RXN", 0);
++ //snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
++
++ /* Add test specific controls */
++ for (i = 0; i < ARRAY_SIZE(mainstone_controls); i++) {
++ if ((err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&mainstone_controls[i],codec, NULL))) < 0)
++ return err;
++ }
++
++ /* Add mainstone specific widgets */
++ for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]);
++ }
++
++ /* set up mainstone specific audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++/* configure the system audio clock */
++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 24576000);
++}
++
++static struct snd_soc_dai_link mainstone_dai[] = {
++{
++ .name = "AC97",
++ .stream_name = "AC97 HiFi",
++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
++ .init = mainstone_wm9713_init,
++ .config_sysclk = mainstone_config_sysclk,
++},
++{
++ .name = "AC97 Aux",
++ .stream_name = "AC97 Aux",
++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
++ .config_sysclk = mainstone_config_sysclk,
++},
++{
++ .name = "WM9713",
++ .stream_name = "WM9713 Voice",
++ .cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2],
++ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
++ .config_sysclk = mainstone_config_sysclk,
++},
++};
++
++static struct snd_soc_machine mainstone = {
++ .name = "Mainstone",
++ .probe = mainstone_probe,
++ .remove = mainstone_remove,
++ .suspend_pre = mainstone_suspend,
++ .resume_post = mainstone_resume,
++ .ops = &mainstone_ops,
++ .dai_link = mainstone_dai,
++ .num_links = ARRAY_SIZE(mainstone_dai),
++};
++
++static struct snd_soc_device mainstone_snd_ac97_devdata = {
++ .machine = &mainstone,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm9713,
++};
++
++static struct platform_device *mainstone_snd_ac97_device;
++
++static int __init mainstone_init(void)
++{
++ int ret;
++
++ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1);
++ if (!mainstone_snd_ac97_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata);
++ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev;
++
++ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0)
++ platform_device_put(mainstone_snd_ac97_device);
++
++ return ret;
++}
++
++static void __exit mainstone_exit(void)
++{
++ platform_device_unregister(mainstone_snd_ac97_device);
++}
++
++module_init(mainstone_init);
++module_exit(mainstone_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM9713 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/poodle.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/poodle.c
+@@ -0,0 +1,329 @@
++/*
++ * poodle.c -- SoC audio for Poodle
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
++ * Richard Purdie <richard@openedhand.com>
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/timer.h>
++#include <linux/interrupt.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/mach-types.h>
++#include <asm/hardware/locomo.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/hardware.h>
++#include <asm/arch/poodle.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/wm8731.h"
++#include "pxa2xx-pcm.h"
++
++#define POODLE_HP 1
++#define POODLE_HP_OFF 0
++#define POODLE_SPK_ON 1
++#define POODLE_SPK_OFF 0
++
++ /* audio clock in Hz - rounded from 12.235MHz */
++#define POODLE_AUDIO_CLOCK 12288000
++
++static int poodle_jack_func;
++static int poodle_spk_func;
++
++static void poodle_ext_control(struct snd_soc_codec *codec)
++{
++ int spk = 0;
++
++ /* set up jack connection */
++ if (poodle_jack_func == POODLE_HP) {
++ /* set = unmute headphone */
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_L, 1);
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_R, 1);
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
++ } else {
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_L, 0);
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_R, 0);
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
++ }
++
++ if (poodle_spk_func == POODLE_SPK_ON)
++ spk = 1;
++
++ /* set the enpoints to their new connetion states */
++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
++
++ /* signal a DAPM event */
++ snd_soc_dapm_sync_endpoints(codec);
++}
++
++static int poodle_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec *codec = rtd->socdev->codec;
++
++ /* check the jack status at stream startup */
++ poodle_ext_control(codec);
++ return 0;
++}
++
++/* we need to unmute the HP at shutdown as the mute burns power on poodle */
++static int poodle_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec *codec = rtd->socdev->codec;
++
++ /* set = unmute headphone */
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_L, 1);
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_R, 1);
++ return 0;
++}
++
++static struct snd_soc_ops poodle_ops = {
++ .startup = poodle_startup,
++ .shutdown = poodle_shutdown,
++};
++
++static int poodle_get_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = poodle_jack_func;
++ return 0;
++}
++
++static int poodle_set_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (poodle_jack_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ poodle_jack_func = ucontrol->value.integer.value[0];
++ poodle_ext_control(codec);
++ return 1;
++}
++
++static int poodle_get_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = poodle_spk_func;
++ return 0;
++}
++
++static int poodle_set_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (poodle_spk_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ poodle_spk_func = ucontrol->value.integer.value[0];
++ poodle_ext_control(codec);
++ return 1;
++}
++
++static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event)
++{
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_AMP_ON, 0);
++ else
++ locomo_gpio_write(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_AMP_ON, 1);
++
++ return 0;
++}
++
++/* poodle machine dapm widgets */
++static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
++SND_SOC_DAPM_HP("Headphone Jack", NULL),
++SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
++};
++
++/* Corgi machine audio_mapnections to the codec pins */
++static const char *audio_map[][3] = {
++
++ /* headphone connected to LHPOUT1, RHPOUT1 */
++ {"Headphone Jack", NULL, "LHPOUT"},
++ {"Headphone Jack", NULL, "RHPOUT"},
++
++ /* speaker connected to LOUT, ROUT */
++ {"Ext Spk", NULL, "ROUT"},
++ {"Ext Spk", NULL, "LOUT"},
++
++ {NULL, NULL, NULL},
++};
++
++static const char *jack_function[] = {"Off", "Headphone"};
++static const char *spk_function[] = {"Off", "On"};
++static const struct soc_enum poodle_enum[] = {
++ SOC_ENUM_SINGLE_EXT(2, jack_function),
++ SOC_ENUM_SINGLE_EXT(2, spk_function),
++};
++
++static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
++ SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
++ poodle_set_jack),
++ SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
++ poodle_set_spk),
++};
++
++/*
++ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
++ */
++static int poodle_wm8731_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
++ snd_soc_dapm_set_endpoint(codec, "MICIN", 1);
++
++ /* Add poodle specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ /* Add poodle specific widgets */
++ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
++ }
++
++ /* Set up poodle specific audio path audio_map */
++ for (i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++static unsigned int poodle_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) {
++ /* pxa2xx is i2s master */
++ switch (info->rate) {
++ case 44100:
++ case 88200:
++ /* configure codec digital filters for 44.1, 88.2 */
++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ 11289600);
++ break;
++ default:
++ /* configure codec digital filters for all other rates */
++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ POODLE_AUDIO_CLOCK);
++ break;
++ }
++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info,
++ POODLE_AUDIO_CLOCK);
++ } else {
++ /* codec is i2s master -
++ * only configure codec DAI clock and filters */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ POODLE_AUDIO_CLOCK);
++ }
++}
++
++/* poodle digital audio interface glue - connects codec <--> CPU */
++static struct snd_soc_dai_link poodle_dai = {
++ .name = "WM8731",
++ .stream_name = "WM8731",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8731_dai,
++ .init = poodle_wm8731_init,
++ .config_sysclk = poodle_config_sysclk,
++};
++
++/* poodle audio machine driver */
++static struct snd_soc_machine snd_soc_machine_poodle = {
++ .name = "Poodle",
++ .dai_link = &poodle_dai,
++ .num_links = 1,
++ .ops = &poodle_ops,
++};
++
++/* poodle audio private data */
++static struct wm8731_setup_data poodle_wm8731_setup = {
++ .i2c_address = 0x1b,
++};
++
++/* poodle audio subsystem */
++static struct snd_soc_device poodle_snd_devdata = {
++ .machine = &snd_soc_machine_poodle,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8731,
++ .codec_data = &poodle_wm8731_setup,
++};
++
++static struct platform_device *poodle_snd_device;
++
++static int __init poodle_init(void)
++{
++ int ret;
++
++ if (!machine_is_poodle())
++ return -ENODEV;
++
++ locomo_gpio_set_dir(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_AMP_ON, 0);
++ /* should we mute HP at startup - burning power ?*/
++ locomo_gpio_set_dir(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_L, 0);
++ locomo_gpio_set_dir(&poodle_locomo_device.dev,
++ POODLE_LOCOMO_GPIO_MUTE_R, 0);
++
++ poodle_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!poodle_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata);
++ poodle_snd_devdata.dev = &poodle_snd_device->dev;
++ ret = platform_device_add(poodle_snd_device);
++
++ if (ret)
++ platform_device_put(poodle_snd_device);
++
++ return ret;
++}
++
++static void __exit poodle_exit(void)
++{
++ platform_device_unregister(poodle_snd_device);
++}
++
++module_init(poodle_init);
++module_exit(poodle_exit);
++
++/* Module information */
++MODULE_AUTHOR("Richard Purdie");
++MODULE_DESCRIPTION("ALSA SoC Poodle");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ac97.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ac97.c
+@@ -0,0 +1,437 @@
++/*
++ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
++ *
++ * Author: Nicolas Pitre
++ * Created: Dec 02, 2004
++ * Copyright: MontaVista Software Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++#include <linux/interrupt.h>
++#include <linux/wait.h>
++#include <linux/delay.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/ac97_codec.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#include <asm/irq.h>
++#include <linux/mutex.h>
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++
++#include "pxa2xx-pcm.h"
++
++static DEFINE_MUTEX(car_mutex);
++static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
++static volatile long gsr_bits;
++
++#define AC97_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define AC97_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
++
++/* may need to expand this */
++static struct snd_soc_dai_mode pxa2xx_ac97_modes[] = {
++ {
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = AC97_RATES,
++ .pcmdir = AC97_DIR,
++ },
++};
++
++/*
++ * Beware PXA27x bugs:
++ *
++ * o Slot 12 read from modem space will hang controller.
++ * o CDONE, SDONE interrupt fails after any slot 12 IO.
++ *
++ * We therefore have an hybrid approach for waiting on SDONE (interrupt or
++ * 1 jiffy timeout if interrupt never comes).
++ */
++
++static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97,
++ unsigned short reg)
++{
++ unsigned short val = -1;
++ volatile u32 *reg_addr;
++
++ mutex_lock(&car_mutex);
++
++ /* set up primary or secondary codec/modem space */
++#ifdef CONFIG_PXA27x
++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
++#else
++ if (reg == AC97_GPIO_STATUS)
++ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
++ else
++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
++#endif
++ reg_addr += (reg >> 1);
++
++#ifndef CONFIG_PXA27x
++ if (reg == AC97_GPIO_STATUS) {
++ /* read from controller cache */
++ val = *reg_addr;
++ goto out;
++ }
++#endif
++
++ /* start read access across the ac97 link */
++ GSR = GSR_CDONE | GSR_SDONE;
++ gsr_bits = 0;
++ val = *reg_addr;
++
++ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
++ if (!((GSR | gsr_bits) & GSR_SDONE)) {
++ printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n",
++ __FUNCTION__, reg, GSR | gsr_bits);
++ val = -1;
++ goto out;
++ }
++
++ /* valid data now */
++ GSR = GSR_CDONE | GSR_SDONE;
++ gsr_bits = 0;
++ val = *reg_addr;
++ /* but we've just started another cycle... */
++ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
++
++out: mutex_unlock(&car_mutex);
++ return val;
++}
++
++static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
++ unsigned short val)
++{
++ volatile u32 *reg_addr;
++
++ mutex_lock(&car_mutex);
++
++ /* set up primary or secondary codec/modem space */
++#ifdef CONFIG_PXA27x
++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
++#else
++ if (reg == AC97_GPIO_STATUS)
++ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
++ else
++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
++#endif
++ reg_addr += (reg >> 1);
++
++ GSR = GSR_CDONE | GSR_SDONE;
++ gsr_bits = 0;
++ *reg_addr = val;
++ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1);
++ if (!((GSR | gsr_bits) & GSR_CDONE))
++ printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n",
++ __FUNCTION__, reg, GSR | gsr_bits);
++
++ mutex_unlock(&car_mutex);
++}
++
++static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
++{
++ gsr_bits = 0;
++
++#ifdef CONFIG_PXA27x
++ /* warm reset broken on Bulverde,
++ so manually keep AC97 reset high */
++ pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH);
++ udelay(10);
++ GCR |= GCR_WARM_RST;
++ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
++ udelay(500);
++#else
++ GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN;
++ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
++#endif
++
++ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
++ printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
++ __FUNCTION__, gsr_bits);
++
++ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
++ GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
++}
++
++static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
++{
++ GCR &= GCR_COLD_RST; /* clear everything but nCRST */
++ GCR &= ~GCR_COLD_RST; /* then assert nCRST */
++
++ gsr_bits = 0;
++#ifdef CONFIG_PXA27x
++ /* PXA27x Developers Manual section 13.5.2.2.1 */
++ pxa_set_cken(1 << 31, 1);
++ udelay(5);
++ pxa_set_cken(1 << 31, 0);
++ GCR = GCR_COLD_RST;
++ udelay(50);
++#else
++ GCR = GCR_COLD_RST;
++ GCR |= GCR_CDONE_IE|GCR_SDONE_IE;
++ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
++#endif
++
++ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
++ printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
++ __FUNCTION__, gsr_bits);
++
++ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
++ GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
++}
++
++static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id)
++{
++ long status;
++
++ status = GSR;
++ if (status) {
++ GSR = status;
++ gsr_bits |= status;
++ wake_up(&gsr_wq);
++
++#ifdef CONFIG_PXA27x
++ /* Although we don't use those we still need to clear them
++ since they tend to spuriously trigger when MMC is used
++ (hardware bug? go figure)... */
++ MISR = MISR_EOC;
++ PISR = PISR_EOC;
++ MCSR = MCSR_EOC;
++#endif
++
++ return IRQ_HANDLED;
++ }
++
++ return IRQ_NONE;
++}
++
++struct snd_ac97_bus_ops soc_ac97_ops = {
++ .read = pxa2xx_ac97_read,
++ .write = pxa2xx_ac97_write,
++ .warm_reset = pxa2xx_ac97_warm_reset,
++ .reset = pxa2xx_ac97_cold_reset,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
++ .name = "AC97 PCM Stereo out",
++ .dev_addr = __PREG(PCDR),
++ .drcmr = &DRCMRTXPCDR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST32 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
++ .name = "AC97 PCM Stereo in",
++ .dev_addr = __PREG(PCDR),
++ .drcmr = &DRCMRRXPCDR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST32 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
++ .name = "AC97 Aux PCM (Slot 5) Mono out",
++ .dev_addr = __PREG(MODR),
++ .drcmr = &DRCMRTXMODR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
++ .name = "AC97 Aux PCM (Slot 5) Mono in",
++ .dev_addr = __PREG(MODR),
++ .drcmr = &DRCMRRXMODR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
++ .name = "AC97 Mic PCM (Slot 6) Mono in",
++ .dev_addr = __PREG(MCDR),
++ .drcmr = &DRCMRRXMCDR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++#ifdef CONFIG_PM
++static int pxa2xx_ac97_suspend(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *dai)
++{
++ GCR |= GCR_ACLINK_OFF;
++ pxa_set_cken(CKEN2_AC97, 0);
++ return 0;
++}
++
++static int pxa2xx_ac97_resume(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *dai)
++{
++ pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
++ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
++ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
++ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
++#ifdef CONFIG_PXA27x
++ /* Use GPIO 113 as AC97 Reset on Bulverde */
++ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
++#endif
++ pxa_set_cken(CKEN2_AC97, 1);
++ return 0;
++}
++
++#else
++#define pxa2xx_ac97_suspend NULL
++#define pxa2xx_ac97_resume NULL
++#endif
++
++static int pxa2xx_ac97_probe(struct platform_device *pdev)
++{
++ int ret;
++
++ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
++ if (ret < 0)
++ goto err;
++
++ pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
++ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
++ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
++ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
++#ifdef CONFIG_PXA27x
++ /* Use GPIO 113 as AC97 Reset on Bulverde */
++ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
++#endif
++ pxa_set_cken(CKEN2_AC97, 1);
++ return 0;
++
++ err:
++ if (CKEN & CKEN2_AC97) {
++ GCR |= GCR_ACLINK_OFF;
++ free_irq(IRQ_AC97, NULL);
++ pxa_set_cken(CKEN2_AC97, 0);
++ }
++ return ret;
++}
++
++static void pxa2xx_ac97_remove(struct platform_device *pdev)
++{
++ GCR |= GCR_ACLINK_OFF;
++ free_irq(IRQ_AC97, NULL);
++ pxa_set_cken(CKEN2_AC97, 0);
++}
++
++static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
++ else
++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
++
++ return 0;
++}
++
++static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
++ else
++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
++
++ return 0;
++}
++
++static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ return -ENODEV;
++ else
++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
++
++ return 0;
++}
++
++/*
++ * There is only 1 physical AC97 interface for pxa2xx, but it
++ * has extra fifo's that can be used for aux DACs and ADCs.
++ */
++struct snd_soc_cpu_dai pxa_ac97_dai[] = {
++{
++ .name = "pxa2xx-ac97",
++ .id = 0,
++ .type = SND_SOC_DAI_AC97,
++ .probe = pxa2xx_ac97_probe,
++ .remove = pxa2xx_ac97_remove,
++ .suspend = pxa2xx_ac97_suspend,
++ .resume = pxa2xx_ac97_resume,
++ .playback = {
++ .stream_name = "AC97 Playback",
++ .channels_min = 2,
++ .channels_max = 2,},
++ .capture = {
++ .stream_name = "AC97 Capture",
++ .channels_min = 2,
++ .channels_max = 2,},
++ .ops = {
++ .hw_params = pxa2xx_ac97_hw_params,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes),
++ .mode = pxa2xx_ac97_modes,},
++},
++{
++ .name = "pxa2xx-ac97-aux",
++ .id = 1,
++ .type = SND_SOC_DAI_AC97,
++ .playback = {
++ .stream_name = "AC97 Aux Playback",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .capture = {
++ .stream_name = "AC97 Aux Capture",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .ops = {
++ .hw_params = pxa2xx_ac97_hw_aux_params,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes),
++ .mode = pxa2xx_ac97_modes,},
++},
++{
++ .name = "pxa2xx-ac97-mic",
++ .id = 2,
++ .type = SND_SOC_DAI_AC97,
++ .capture = {
++ .stream_name = "AC97 Mic Capture",
++ .channels_min = 1,
++ .channels_max = 1,},
++ .ops = {
++ .hw_params = pxa2xx_ac97_hw_mic_params,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes),
++ .mode = pxa2xx_ac97_modes,},},
++};
++
++EXPORT_SYMBOL_GPL(pxa_ac97_dai);
++EXPORT_SYMBOL_GPL(soc_ac97_ops);
++
++MODULE_AUTHOR("Nicolas Pitre");
++MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-i2s.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-i2s.c
+@@ -0,0 +1,354 @@
++/*
++ * pxa2xx-i2s.c -- ALSA Soc Audio Layer
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 12th Aug 2005 Initial version.
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/device.h>
++#include <linux/delay.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++
++#include "pxa2xx-pcm.h"
++
++/* used to disable sysclk if external crystal is used */
++static int extclk;
++module_param(extclk, int, 0);
++MODULE_PARM_DESC(extclk, "set to 1 to disable pxa2xx i2s sysclk");
++
++struct pxa_i2s_port {
++ u32 sadiv;
++ u32 sacr0;
++ u32 sacr1;
++ u32 saimr;
++ int master;
++};
++static struct pxa_i2s_port pxa_i2s;
++
++#define PXA_I2S_DAIFMT \
++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF)
++
++#define PXA_I2S_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define PXA_I2S_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++/* priv is divider */
++static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = {
++ /* pxa2xx I2S frame and clock master modes */
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x48,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x34,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x24,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0x1a,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0xd,
++ },
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = PXA_I2S_DIR,
++ .flags = SND_SOC_DAI_BFS_DIV,
++ .fs = 256,
++ .bfs = SND_SOC_FSBD(4),
++ .priv = 0xc,
++ },
++
++ /* pxa2xx I2S frame master and clock slave mode */
++ {
++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS,
++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++ .pcmrate = PXA_I2S_RATES,
++ .pcmdir = PXA_I2S_DIR,
++ .fs = SND_SOC_FS_ALL,
++ .flags = SND_SOC_DAI_BFS_RATE,
++ .bfs = 64,
++ .priv = 0x48,
++ },
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
++ .name = "I2S PCM Stereo out",
++ .dev_addr = __PREG(SADR),
++ .drcmr = &DRCMRTXSADR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST32 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
++ .name = "I2S PCM Stereo in",
++ .dev_addr = __PREG(SADR),
++ .drcmr = &DRCMRRXSADR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST32 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_gpio gpio_bus[] = {
++ { /* I2S SoC Slave */
++ .rx = GPIO29_SDATA_IN_I2S_MD,
++ .tx = GPIO30_SDATA_OUT_I2S_MD,
++ .clk = GPIO28_BITCLK_IN_I2S_MD,
++ .frm = GPIO31_SYNC_I2S_MD,
++ },
++ { /* I2S SoC Master */
++#ifdef CONFIG_PXA27x
++ .sys = GPIO113_I2S_SYSCLK_MD,
++#else
++ .sys = GPIO32_SYSCLK_I2S_MD,
++#endif
++ .rx = GPIO29_SDATA_IN_I2S_MD,
++ .tx = GPIO30_SDATA_OUT_I2S_MD,
++ .clk = GPIO28_BITCLK_OUT_I2S_MD,
++ .frm = GPIO31_SYNC_I2S_MD,
++ },
++};
++
++static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if (!rtd->cpu_dai->active) {
++ SACR0 |= SACR0_RST;
++ SACR0 = 0;
++ }
++
++ return 0;
++}
++
++/* wait for I2S controller to be ready */
++static int pxa_i2s_wait(void)
++{
++ int i;
++
++ /* flush the Rx FIFO */
++ for(i = 0; i < 16; i++)
++ SADR;
++ return 0;
++}
++
++static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ pxa_i2s.master = 0;
++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CBS_CFS)
++ pxa_i2s.master = 1;
++
++ if (pxa_i2s.master && !extclk)
++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
++
++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
++ pxa_set_cken(CKEN8_I2S, 1);
++ pxa_i2s_wait();
++
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ rtd->cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
++ else
++ rtd->cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
++
++ /* is port used by another stream */
++ if (!(SACR0 & SACR0_ENB)) {
++
++ SACR0 = 0;
++ SACR1 = 0;
++ if (pxa_i2s.master)
++ SACR0 |= SACR0_BCKD;
++
++ SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1);
++
++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_LEFT_J)
++ SACR1 |= SACR1_AMSL;
++ }
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ SAIMR |= SAIMR_TFS;
++ else
++ SAIMR |= SAIMR_RFS;
++
++ SADIV = rtd->cpu_dai->dai_runtime.priv;
++ return 0;
++}
++
++static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++ int ret = 0;
++
++ switch (cmd) {
++ case SNDRV_PCM_TRIGGER_START:
++ SACR0 |= SACR0_ENB;
++ break;
++ case SNDRV_PCM_TRIGGER_RESUME:
++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++ case SNDRV_PCM_TRIGGER_STOP:
++ case SNDRV_PCM_TRIGGER_SUSPEND:
++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++ break;
++ default:
++ ret = -EINVAL;
++ }
++
++ return ret;
++}
++
++static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
++{
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++ SACR1 |= SACR1_DRPL;
++ SAIMR &= ~SAIMR_TFS;
++ } else {
++ SACR1 |= SACR1_DREC;
++ SAIMR &= ~SAIMR_RFS;
++ }
++
++ if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
++ SACR0 &= ~SACR0_ENB;
++ pxa_i2s_wait();
++ pxa_set_cken(CKEN8_I2S, 0);
++ }
++}
++
++#ifdef CONFIG_PM
++static int pxa2xx_i2s_suspend(struct platform_device *dev,
++ struct snd_soc_cpu_dai *dai)
++{
++ if (!dai->active)
++ return 0;
++
++ /* store registers */
++ pxa_i2s.sacr0 = SACR0;
++ pxa_i2s.sacr1 = SACR1;
++ pxa_i2s.saimr = SAIMR;
++ pxa_i2s.sadiv = SADIV;
++
++ /* deactivate link */
++ SACR0 &= ~SACR0_ENB;
++ pxa_i2s_wait();
++ return 0;
++}
++
++static int pxa2xx_i2s_resume(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *dai)
++{
++ if (!dai->active)
++ return 0;
++
++ pxa_i2s_wait();
++
++ SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
++ SACR1 = pxa_i2s.sacr1;
++ SAIMR = pxa_i2s.saimr;
++ SADIV = pxa_i2s.sadiv;
++ SACR0 |= SACR0_ENB;
++
++ return 0;
++}
++
++#else
++#define pxa2xx_i2s_suspend NULL
++#define pxa2xx_i2s_resume NULL
++#endif
++
++/* pxa2xx I2S sysclock is always 256 FS */
++static unsigned int pxa_i2s_config_sysclk(struct snd_soc_cpu_dai *iface,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ return info->rate << 8;
++}
++
++struct snd_soc_cpu_dai pxa_i2s_dai = {
++ .name = "pxa2xx-i2s",
++ .id = 0,
++ .type = SND_SOC_DAI_I2S,
++ .suspend = pxa2xx_i2s_suspend,
++ .resume = pxa2xx_i2s_resume,
++ .config_sysclk = pxa_i2s_config_sysclk,
++ .playback = {
++ .channels_min = 2,
++ .channels_max = 2,},
++ .capture = {
++ .channels_min = 2,
++ .channels_max = 2,},
++ .ops = {
++ .startup = pxa2xx_i2s_startup,
++ .shutdown = pxa2xx_i2s_shutdown,
++ .trigger = pxa2xx_i2s_trigger,
++ .hw_params = pxa2xx_i2s_hw_params,},
++ .caps = {
++ .num_modes = ARRAY_SIZE(pxa2xx_i2s_modes),
++ .mode = pxa2xx_i2s_modes,},
++};
++
++EXPORT_SYMBOL_GPL(pxa_i2s_dai);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.c
+@@ -0,0 +1,363 @@
++/*
++ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
++ *
++ * Author: Nicolas Pitre
++ * Created: Nov 30, 2004
++ * Copyright: (C) 2004 MontaVista Software, Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/init.h>
++#include <linux/platform_device.h>
++#include <linux/slab.h>
++#include <linux/dma-mapping.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++
++#include <asm/dma.h>
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++
++#include "pxa2xx-pcm.h"
++
++static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
++ .info = SNDRV_PCM_INFO_MMAP |
++ SNDRV_PCM_INFO_MMAP_VALID |
++ SNDRV_PCM_INFO_INTERLEAVED |
++ SNDRV_PCM_INFO_PAUSE |
++ SNDRV_PCM_INFO_RESUME,
++ .formats = SNDRV_PCM_FMTBIT_S16_LE |
++ SNDRV_PCM_FMTBIT_S24_LE |
++ SNDRV_PCM_FMTBIT_S32_LE,
++ .period_bytes_min = 32,
++ .period_bytes_max = 8192 - 32,
++ .periods_min = 1,
++ .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc),
++ .buffer_bytes_max = 128 * 1024,
++ .fifo_size = 32,
++};
++
++struct pxa2xx_runtime_data {
++ int dma_ch;
++ struct pxa2xx_pcm_dma_params *params;
++ pxa_dma_desc *dma_desc_array;
++ dma_addr_t dma_desc_array_phys;
++};
++
++static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
++{
++ struct snd_pcm_substream *substream = dev_id;
++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
++ int dcsr;
++
++ dcsr = DCSR(dma_ch);
++ DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
++
++ if (dcsr & DCSR_ENDINTR) {
++ snd_pcm_period_elapsed(substream);
++ } else {
++ printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
++ prtd->params->name, dma_ch, dcsr );
++ }
++}
++
++static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct pxa2xx_runtime_data *prtd = runtime->private_data;
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct pxa2xx_pcm_dma_params *dma = rtd->cpu_dai->dma_data;
++ size_t totsize = params_buffer_bytes(params);
++ size_t period = params_period_bytes(params);
++ pxa_dma_desc *dma_desc;
++ dma_addr_t dma_buff_phys, next_desc_phys;
++ int ret;
++
++ /* this may get called several times by oss emulation
++ * with different params */
++ if (prtd->params == NULL) {
++ prtd->params = dma;
++ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
++ pxa2xx_pcm_dma_irq, substream);
++ if (ret < 0)
++ return ret;
++ prtd->dma_ch = ret;
++ } else if (prtd->params != dma) {
++ pxa_free_dma(prtd->dma_ch);
++ prtd->params = dma;
++ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
++ pxa2xx_pcm_dma_irq, substream);
++ if (ret < 0)
++ return ret;
++ prtd->dma_ch = ret;
++ }
++
++ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
++ runtime->dma_bytes = totsize;
++
++ dma_desc = prtd->dma_desc_array;
++ next_desc_phys = prtd->dma_desc_array_phys;
++ dma_buff_phys = runtime->dma_addr;
++ do {
++ next_desc_phys += sizeof(pxa_dma_desc);
++ dma_desc->ddadr = next_desc_phys;
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++ dma_desc->dsadr = dma_buff_phys;
++ dma_desc->dtadr = prtd->params->dev_addr;
++ } else {
++ dma_desc->dsadr = prtd->params->dev_addr;
++ dma_desc->dtadr = dma_buff_phys;
++ }
++ if (period > totsize)
++ period = totsize;
++ dma_desc->dcmd = prtd->params->dcmd | period | DCMD_ENDIRQEN;
++ dma_desc++;
++ dma_buff_phys += period;
++ } while (totsize -= period);
++ dma_desc[-1].ddadr = prtd->dma_desc_array_phys;
++
++ return 0;
++}
++
++static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
++
++ if (prtd && prtd->params)
++ *prtd->params->drcmr = 0;
++
++ if (prtd->dma_ch) {
++ snd_pcm_set_runtime_buffer(substream, NULL);
++ pxa_free_dma(prtd->dma_ch);
++ prtd->dma_ch = 0;
++ }
++
++ return 0;
++}
++
++static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
++{
++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
++
++ DCSR(prtd->dma_ch) &= ~DCSR_RUN;
++ DCSR(prtd->dma_ch) = 0;
++ DCMD(prtd->dma_ch) = 0;
++ *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
++
++ return 0;
++}
++
++static int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
++ int ret = 0;
++
++ switch (cmd) {
++ case SNDRV_PCM_TRIGGER_START:
++ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
++ DCSR(prtd->dma_ch) = DCSR_RUN;
++ break;
++
++ case SNDRV_PCM_TRIGGER_STOP:
++ case SNDRV_PCM_TRIGGER_SUSPEND:
++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++ DCSR(prtd->dma_ch) &= ~DCSR_RUN;
++ break;
++
++ case SNDRV_PCM_TRIGGER_RESUME:
++ DCSR(prtd->dma_ch) |= DCSR_RUN;
++ break;
++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
++ DCSR(prtd->dma_ch) |= DCSR_RUN;
++ break;
++
++ default:
++ ret = -EINVAL;
++ }
++
++ return ret;
++}
++
++static snd_pcm_uframes_t
++pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct pxa2xx_runtime_data *prtd = runtime->private_data;
++
++ dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
++ DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
++ snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
++
++ if (x == runtime->buffer_size)
++ x = 0;
++ return x;
++}
++
++static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct pxa2xx_runtime_data *prtd;
++ int ret;
++
++ snd_soc_set_runtime_hwparams(substream, &pxa2xx_pcm_hardware);
++
++ /*
++ * For mysterious reasons (and despite what the manual says)
++ * playback samples are lost if the DMA count is not a multiple
++ * of the DMA burst size. Let's add a rule to enforce that.
++ */
++ ret = snd_pcm_hw_constraint_step(runtime, 0,
++ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
++ if (ret)
++ goto out;
++
++ ret = snd_pcm_hw_constraint_step(runtime, 0,
++ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
++ if (ret)
++ goto out;
++
++ prtd = kzalloc(sizeof(struct pxa2xx_runtime_data), GFP_KERNEL);
++ if (prtd == NULL) {
++ ret = -ENOMEM;
++ goto out;
++ }
++
++ prtd->dma_desc_array =
++ dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
++ &prtd->dma_desc_array_phys, GFP_KERNEL);
++ if (!prtd->dma_desc_array) {
++ ret = -ENOMEM;
++ goto err1;
++ }
++
++ runtime->private_data = prtd;
++ return 0;
++
++ err1:
++ kfree(prtd);
++ out:
++ return ret;
++}
++
++static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ struct pxa2xx_runtime_data *prtd = runtime->private_data;
++
++ dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
++ prtd->dma_desc_array, prtd->dma_desc_array_phys);
++ kfree(prtd);
++ return 0;
++}
++
++static int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
++ struct vm_area_struct *vma)
++{
++ struct snd_pcm_runtime *runtime = substream->runtime;
++ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
++ runtime->dma_area,
++ runtime->dma_addr,
++ runtime->dma_bytes);
++}
++
++struct snd_pcm_ops pxa2xx_pcm_ops = {
++ .open = pxa2xx_pcm_open,
++ .close = pxa2xx_pcm_close,
++ .ioctl = snd_pcm_lib_ioctl,
++ .hw_params = pxa2xx_pcm_hw_params,
++ .hw_free = pxa2xx_pcm_hw_free,
++ .prepare = pxa2xx_pcm_prepare,
++ .trigger = pxa2xx_pcm_trigger,
++ .pointer = pxa2xx_pcm_pointer,
++ .mmap = pxa2xx_pcm_mmap,
++};
++
++static int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
++{
++ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
++ struct snd_dma_buffer *buf = &substream->dma_buffer;
++ size_t size = pxa2xx_pcm_hardware.buffer_bytes_max;
++ buf->dev.type = SNDRV_DMA_TYPE_DEV;
++ buf->dev.dev = pcm->card->dev;
++ buf->private_data = NULL;
++ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
++ &buf->addr, GFP_KERNEL);
++ if (!buf->area)
++ return -ENOMEM;
++ buf->bytes = size;
++ return 0;
++}
++
++static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
++{
++ struct snd_pcm_substream *substream;
++ struct snd_dma_buffer *buf;
++ int stream;
++
++ for (stream = 0; stream < 2; stream++) {
++ substream = pcm->streams[stream].substream;
++ if (!substream)
++ continue;
++
++ buf = &substream->dma_buffer;
++ if (!buf->area)
++ continue;
++
++ dma_free_writecombine(pcm->card->dev, buf->bytes,
++ buf->area, buf->addr);
++ buf->area = NULL;
++ }
++}
++
++static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
++
++int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
++ struct snd_pcm *pcm)
++{
++ int ret = 0;
++
++ if (!card->dev->dma_mask)
++ card->dev->dma_mask = &pxa2xx_pcm_dmamask;
++ if (!card->dev->coherent_dma_mask)
++ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
++
++ if (dai->playback.channels_min) {
++ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
++ SNDRV_PCM_STREAM_PLAYBACK);
++ if (ret)
++ goto out;
++ }
++
++ if (dai->capture.channels_min) {
++ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
++ SNDRV_PCM_STREAM_CAPTURE);
++ if (ret)
++ goto out;
++ }
++ out:
++ return ret;
++}
++
++struct snd_soc_platform pxa2xx_soc_platform = {
++ .name = "pxa2xx-audio",
++ .pcm_ops = &pxa2xx_pcm_ops,
++ .pcm_new = pxa2xx_pcm_new,
++ .pcm_free = pxa2xx_pcm_free_dma_buffers,
++};
++
++EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
++
++MODULE_AUTHOR("Nicolas Pitre");
++MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.h
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.h
+@@ -0,0 +1,48 @@
++/*
++ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
++ *
++ * Author: Nicolas Pitre
++ * Created: Nov 30, 2004
++ * Copyright: MontaVista Software, Inc.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef _PXA2XX_PCM_H
++#define _PXA2XX_PCM_H
++
++struct pxa2xx_pcm_dma_params {
++ char *name; /* stream identifier */
++ u32 dcmd; /* DMA descriptor dcmd field */
++ volatile u32 *drcmr; /* the DMA request channel to use */
++ u32 dev_addr; /* device physical address for DMA */
++};
++
++struct pxa2xx_gpio {
++ u32 sys;
++ u32 rx;
++ u32 tx;
++ u32 clk;
++ u32 frm;
++};
++
++/* pxa2xx DAI ID's */
++#define PXA2XX_DAI_AC97_HIFI 0
++#define PXA2XX_DAI_AC97_AUX 1
++#define PXA2XX_DAI_AC97_MIC 2
++#define PXA2XX_DAI_I2S 0
++#define PXA2XX_DAI_SSP1 0
++#define PXA2XX_DAI_SSP2 1
++#define PXA2XX_DAI_SSP3 2
++
++extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
++extern struct snd_soc_cpu_dai pxa_i2s_dai;
++extern struct snd_soc_cpu_dai pxa_ssp_dai[3];
++
++/* platform data */
++extern struct snd_soc_platform pxa2xx_soc_platform;
++extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
++
++#endif
+Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ssp.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ssp.c
+@@ -0,0 +1,767 @@
++/*
++ * pxa2xx-ssp.c -- ALSA Soc Audio Layer
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood@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
++ * 12th Aug 2005 Initial version.
++ *
++ * TODO:
++ * o Fix master mode (bug)
++ * o Fix resume (bug)
++ * o Add support for other clocks
++ * o Test network mode for > 16bit sample size
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++#include <asm/arch/ssp.h>
++
++#include "pxa2xx-pcm.h"
++
++/*
++ * SSP sysclock frequency in Hz
++ * Neither default pxa2xx PLL clocks are good for audio, hence pxa27x
++ * has audio clock. I would recommend using the pxa27x audio clock or an
++ * external clock or making the codec master to gurantee better sample rates.
++ */
++#ifdef CONFIG_PXA27x
++static int sysclk[3] = {13000000, 13000000, 13000000};
++#else
++static int sysclk[3] = {1843200, 1843200, 1843200};
++#endif
++module_param_array(sysclk, int, NULL, 0);
++MODULE_PARM_DESC(sysclk, "sysclk frequency in Hz");
++
++/*
++ * SSP sysclock source.
++ * sysclk is ignored if audio clock is used
++ */
++#ifdef CONFIG_PXA27x
++static int clksrc[3] = {0, 0, 0};
++#else
++static int clksrc[3] = {0, 0, 0};
++#endif
++module_param_array(clksrc, int, NULL, 0);
++MODULE_PARM_DESC(clksrc,
++ "sysclk source, 0 = internal PLL, 1 = ext, 2 = network, 3 = audio clock");
++
++/*
++ * SSP GPIO's
++ */
++#define GPIO26_SSP1RX_MD (26 | GPIO_ALT_FN_1_IN)
++#define GPIO25_SSP1TX_MD (25 | GPIO_ALT_FN_2_OUT)
++#define GPIO23_SSP1CLKS_MD (23 | GPIO_ALT_FN_2_IN)
++#define GPIO24_SSP1FRMS_MD (24 | GPIO_ALT_FN_2_IN)
++#define GPIO23_SSP1CLKM_MD (23 | GPIO_ALT_FN_2_OUT)
++#define GPIO24_SSP1FRMM_MD (24 | GPIO_ALT_FN_2_OUT)
++
++#define GPIO11_SSP2RX_MD (11 | GPIO_ALT_FN_2_IN)
++#define GPIO13_SSP2TX_MD (13 | GPIO_ALT_FN_1_OUT)
++#define GPIO22_SSP2CLKS_MD (22 | GPIO_ALT_FN_3_IN)
++#define GPIO88_SSP2FRMS_MD (88 | GPIO_ALT_FN_3_IN)
++#define GPIO22_SSP2CLKM_MD (22 | GPIO_ALT_FN_3_OUT)
++#define GPIO88_SSP2FRMM_MD (88 | GPIO_ALT_FN_3_OUT)
++
++#define GPIO82_SSP3RX_MD (82 | GPIO_ALT_FN_1_IN)
++#define GPIO81_SSP3TX_MD (81 | GPIO_ALT_FN_1_OUT)
++#define GPIO84_SSP3CLKS_MD (84 | GPIO_ALT_FN_1_IN)
++#define GPIO83_SSP3FRMS_MD (83 | GPIO_ALT_FN_1_IN)
++#define GPIO84_SSP3CLKM_MD (84 | GPIO_ALT_FN_1_OUT)
++#define GPIO83_SSP3FRMM_MD (83 | GPIO_ALT_FN_1_OUT)
++
++#define PXA_SSP_MDAIFMT \
++ (SND_SOC_DAIFMT_DSP_B |SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_CBM_CFS | \
++ SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF)
++
++#define PXA_SSP_SDAIFMT \
++ (SND_SOC_DAIFMT_DSP_B |SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS | \
++ SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF)
++
++#define PXA_SSP_DIR \
++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++#define PXA_SSP_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
++ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
++
++#define PXA_SSP_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++/*
++ * SSP modes
++ */
++static struct snd_soc_dai_mode pxa2xx_ssp_modes[] = {
++ /* port slave clk & frame modes */
++ {
++ .fmt = PXA_SSP_SDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = PXA_SSP_RATES,
++ .pcmdir = PXA_SSP_DIR,
++ .fs = SND_SOC_FS_ALL,
++ .bfs = SND_SOC_FSB_ALL,
++ },
++
++ /* port master clk & frame modes */
++#ifdef CONFIG_PXA27x
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_8000,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_11025,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_16000,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_22050,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_32000,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_44100,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_48000,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 256,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_88200,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 128,
++ .bfs = SND_SOC_FSBW(1),
++ },
++ {
++ .fmt = PXA_SSP_MDAIFMT,
++ .pcmfmt = PXA_SSP_BITS,
++ .pcmrate = SNDRV_PCM_RATE_96000,
++ .pcmdir = PXA_SSP_DIR,
++ .flags = SND_SOC_DAI_BFS_RCW,
++ .fs = 128,
++ .bfs = SND_SOC_FSBW(1),
++ },
++#endif
++};
++
++static struct ssp_dev ssp[3];
++#ifdef CONFIG_PM
++static struct ssp_state ssp_state[3];
++#endif
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_out = {
++ .name = "SSP1 PCM Mono out",
++ .dev_addr = __PREG(SSDR_P1),
++ .drcmr = &DRCMRTXSSDR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_in = {
++ .name = "SSP1 PCM Mono in",
++ .dev_addr = __PREG(SSDR_P1),
++ .drcmr = &DRCMRRXSSDR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_out = {
++ .name = "SSP1 PCM Stereo out",
++ .dev_addr = __PREG(SSDR_P1),
++ .drcmr = &DRCMRTXSSDR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_in = {
++ .name = "SSP1 PCM Stereo in",
++ .dev_addr = __PREG(SSDR_P1),
++ .drcmr = &DRCMRRXSSDR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_out = {
++ .name = "SSP2 PCM Mono out",
++ .dev_addr = __PREG(SSDR_P2),
++ .drcmr = &DRCMRTXSS2DR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_in = {
++ .name = "SSP2 PCM Mono in",
++ .dev_addr = __PREG(SSDR_P2),
++ .drcmr = &DRCMRRXSS2DR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_out = {
++ .name = "SSP2 PCM Stereo out",
++ .dev_addr = __PREG(SSDR_P2),
++ .drcmr = &DRCMRTXSS2DR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_in = {
++ .name = "SSP2 PCM Stereo in",
++ .dev_addr = __PREG(SSDR_P2),
++ .drcmr = &DRCMRRXSS2DR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_out = {
++ .name = "SSP3 PCM Mono out",
++ .dev_addr = __PREG(SSDR_P3),
++ .drcmr = &DRCMRTXSS3DR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_in = {
++ .name = "SSP3 PCM Mono in",
++ .dev_addr = __PREG(SSDR_P3),
++ .drcmr = &DRCMRRXSS3DR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH2,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_out = {
++ .name = "SSP3 PCM Stereo out",
++ .dev_addr = __PREG(SSDR_P3),
++ .drcmr = &DRCMRTXSS3DR,
++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
++ DCMD_BURST16 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_in = {
++ .name = "SSP3 PCM Stereo in",
++ .dev_addr = __PREG(SSDR_P3),
++ .drcmr = &DRCMRRXSS3DR,
++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
++ DCMD_BURST16 | DCMD_WIDTH4,
++};
++
++static struct pxa2xx_pcm_dma_params *ssp_dma_params[3][4] = {
++ {&pxa2xx_ssp1_pcm_mono_out, &pxa2xx_ssp1_pcm_mono_in,
++ &pxa2xx_ssp1_pcm_stereo_out,&pxa2xx_ssp1_pcm_stereo_in,},
++ {&pxa2xx_ssp2_pcm_mono_out, &pxa2xx_ssp2_pcm_mono_in,
++ &pxa2xx_ssp2_pcm_stereo_out, &pxa2xx_ssp2_pcm_stereo_in,},
++ {&pxa2xx_ssp3_pcm_mono_out, &pxa2xx_ssp3_pcm_mono_in,
++ &pxa2xx_ssp3_pcm_stereo_out,&pxa2xx_ssp3_pcm_stereo_in,},
++};
++
++static struct pxa2xx_gpio ssp_gpios[3][4] = {
++ {{ /* SSP1 SND_SOC_DAIFMT_CBM_CFM */
++ .rx = GPIO26_SSP1RX_MD,
++ .tx = GPIO25_SSP1TX_MD,
++ .clk = (23 | GPIO_ALT_FN_2_IN),
++ .frm = (24 | GPIO_ALT_FN_2_IN),
++ },
++ { /* SSP1 SND_SOC_DAIFMT_CBS_CFS */
++ .rx = GPIO26_SSP1RX_MD,
++ .tx = GPIO25_SSP1TX_MD,
++ .clk = (23 | GPIO_ALT_FN_2_OUT),
++ .frm = (24 | GPIO_ALT_FN_2_OUT),
++ },
++ { /* SSP1 SND_SOC_DAIFMT_CBS_CFM */
++ .rx = GPIO26_SSP1RX_MD,
++ .tx = GPIO25_SSP1TX_MD,
++ .clk = (23 | GPIO_ALT_FN_2_OUT),
++ .frm = (24 | GPIO_ALT_FN_2_IN),
++ },
++ { /* SSP1 SND_SOC_DAIFMT_CBM_CFS */
++ .rx = GPIO26_SSP1RX_MD,
++ .tx = GPIO25_SSP1TX_MD,
++ .clk = (23 | GPIO_ALT_FN_2_IN),
++ .frm = (24 | GPIO_ALT_FN_2_OUT),
++ }},
++ {{ /* SSP2 SND_SOC_DAIFMT_CBM_CFM */
++ .rx = GPIO11_SSP2RX_MD,
++ .tx = GPIO13_SSP2TX_MD,
++ .clk = (22 | GPIO_ALT_FN_3_IN),
++ .frm = (88 | GPIO_ALT_FN_3_IN),
++ },
++ { /* SSP2 SND_SOC_DAIFMT_CBS_CFS */
++ .rx = GPIO11_SSP2RX_MD,
++ .tx = GPIO13_SSP2TX_MD,
++ .clk = (22 | GPIO_ALT_FN_3_OUT),
++ .frm = (88 | GPIO_ALT_FN_3_OUT),
++ },
++ { /* SSP2 SND_SOC_DAIFMT_CBS_CFM */
++ .rx = GPIO11_SSP2RX_MD,
++ .tx = GPIO13_SSP2TX_MD,
++ .clk = (22 | GPIO_ALT_FN_3_OUT),
++ .frm = (88 | GPIO_ALT_FN_3_IN),
++ },
++ { /* SSP2 SND_SOC_DAIFMT_CBM_CFS */
++ .rx = GPIO11_SSP2RX_MD,
++ .tx = GPIO13_SSP2TX_MD,
++ .clk = (22 | GPIO_ALT_FN_3_IN),
++ .frm = (88 | GPIO_ALT_FN_3_OUT),
++ }},
++ {{ /* SSP3 SND_SOC_DAIFMT_CBM_CFM */
++ .rx = GPIO82_SSP3RX_MD,
++ .tx = GPIO81_SSP3TX_MD,
++ .clk = (84 | GPIO_ALT_FN_3_IN),
++ .frm = (83 | GPIO_ALT_FN_3_IN),
++ },
++ { /* SSP3 SND_SOC_DAIFMT_CBS_CFS */
++ .rx = GPIO82_SSP3RX_MD,
++ .tx = GPIO81_SSP3TX_MD,
++ .clk = (84 | GPIO_ALT_FN_3_OUT),
++ .frm = (83 | GPIO_ALT_FN_3_OUT),
++ },
++ { /* SSP3 SND_SOC_DAIFMT_CBS_CFM */
++ .rx = GPIO82_SSP3RX_MD,
++ .tx = GPIO81_SSP3TX_MD,
++ .clk = (84 | GPIO_ALT_FN_3_OUT),
++ .frm = (83 | GPIO_ALT_FN_3_IN),
++ },
++ { /* SSP3 SND_SOC_DAIFMT_CBM_CFS */
++ .rx = GPIO82_SSP3RX_MD,
++ .tx = GPIO81_SSP3TX_MD,
++ .clk = (84 | GPIO_ALT_FN_3_IN),
++ .frm = (83 | GPIO_ALT_FN_3_OUT),
++ }},
++};
++
++static int pxa2xx_ssp_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ int ret = 0;
++
++ if (!rtd->cpu_dai->active) {
++ ret = ssp_init (&ssp[rtd->cpu_dai->id], rtd->cpu_dai->id + 1,
++ SSP_NO_IRQ);
++ if (ret < 0)
++ return ret;
++ ssp_disable(&ssp[rtd->cpu_dai->id]);
++ }
++ return ret;
++}
++
++static void pxa2xx_ssp_shutdown(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++ if (!rtd->cpu_dai->active) {
++ ssp_disable(&ssp[rtd->cpu_dai->id]);
++ ssp_exit(&ssp[rtd->cpu_dai->id]);
++ }
++}
++
++#ifdef CONFIG_PM
++
++#if defined (CONFIG_PXA27x)
++static int cken[3] = {CKEN23_SSP1, CKEN3_SSP2, CKEN4_SSP3};
++#else
++static int cken[3] = {CKEN3_SSP, CKEN9_NSSP, CKEN10_ASSP};
++#endif
++
++static int pxa2xx_ssp_suspend(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *dai)
++{
++ if (!dai->active)
++ return 0;
++
++ ssp_save_state(&ssp[dai->id], &ssp_state[dai->id]);
++ pxa_set_cken(cken[dai->id], 0);
++ return 0;
++}
++
++static int pxa2xx_ssp_resume(struct platform_device *pdev,
++ struct snd_soc_cpu_dai *dai)
++{
++ if (!dai->active)
++ return 0;
++
++ pxa_set_cken(cken[dai->id], 1);
++ ssp_restore_state(&ssp[dai->id], &ssp_state[dai->id]);
++ ssp_enable(&ssp[dai->id]);
++
++ return 0;
++}
++
++#else
++#define pxa2xx_ssp_suspend NULL
++#define pxa2xx_ssp_resume NULL
++#endif
++
++/* todo - check clk source and PLL before returning clock rate */
++static unsigned int pxa_ssp_config_sysclk(struct snd_soc_cpu_dai *dai,
++ struct snd_soc_clock_info *info, unsigned int clk)
++{
++ /* audio clock ? (divide by 1) */
++ if (clksrc[dai->id] == 3) {
++ switch(info->rate){
++ case 8000:
++ case 16000:
++ case 32000:
++ case 48000:
++ case 96000:
++ return 12288000;
++ break;
++ case 11025:
++ case 22050:
++ case 44100:
++ case 88200:
++ return 11289600;
++ break;
++ }
++ }
++
++ /* pll */
++ return sysclk[dai->id];
++}
++
++#ifdef CONFIG_PXA27x
++static u32 pxa27x_set_audio_clk(unsigned int rate, unsigned int fs)
++{
++ u32 aclk = 0, div = 0;
++
++ if (rate == 0 || fs == 0)
++ return 0;
++
++ switch(rate){
++ case 8000:
++ case 16000:
++ case 32000:
++ case 48000:
++ case 96000:
++ aclk = 0x2 << 4;
++ div = 12288000 / (rate * fs);
++ break;
++ case 11025:
++ case 22050:
++ case 44100:
++ case 88200:
++ aclk = 0x1 << 4;
++ div = 11289600 / (rate * fs);
++ break;
++ }
++
++ aclk |= ffs(div) - 1;
++ return aclk;
++}
++#endif
++
++static inline int get_scr(int srate, int id)
++{
++ if (srate == 0)
++ return 0;
++ return (sysclk[id] / srate) - 1;
++}
++
++static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ int fmt = 0, dma = 0, fs, chn = params_channels(params);
++ u32 ssp_mode = 0, ssp_setup = 0, psp_mode = 0, rate = 0;
++
++ fs = rtd->cpu_dai->dai_runtime.fs;
++
++ /* select correct DMA params */
++ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
++ dma = 1;
++ if (chn == 2 || rtd->cpu_dai->dai_runtime.pcmfmt != PXA_SSP_BITS)
++ dma += 2;
++ rtd->cpu_dai->dma_data = ssp_dma_params[rtd->cpu_dai->id][dma];
++
++ /* is port used by another stream */
++ if (SSCR0 & SSCR0_SSE)
++ return 0;
++
++ /* bit size */
++ switch(rtd->cpu_dai->dai_runtime.pcmfmt) {
++ case SNDRV_PCM_FMTBIT_S16_LE:
++ ssp_mode |=SSCR0_DataSize(16);
++ break;
++ case SNDRV_PCM_FMTBIT_S24_LE:
++ ssp_mode |=(SSCR0_EDSS | SSCR0_DataSize(8));
++ /* use network mode for stereo samples > 16 bits */
++ if (chn == 2) {
++ ssp_mode |= (SSCR0_MOD | SSCR0_SlotsPerFrm(2) << 24);
++ /* active slots 0,1 */
++ SSTSA_P(rtd->cpu_dai->id +1) = 0x3;
++ SSRSA_P(rtd->cpu_dai->id +1) = 0x3;
++ }
++ break;
++ case SNDRV_PCM_FMTBIT_S32_LE:
++ ssp_mode |= (SSCR0_EDSS | SSCR0_DataSize(16));
++ /* use network mode for stereo samples > 16 bits */
++ if (chn == 2) {
++ ssp_mode |= (SSCR0_MOD | SSCR0_SlotsPerFrm(2) << 24);
++ /* active slots 0,1 */
++ SSTSA_P(rtd->cpu_dai->id +1) = 0x3;
++ SSRSA_P(rtd->cpu_dai->id +1) = 0x3;
++ }
++ break;
++ }
++
++ ssp_mode |= SSCR0_PSP;
++ ssp_setup = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) |
++ SSCR1_TRAIL | SSCR1_RWOT;
++
++ switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++ case SND_SOC_DAIFMT_CBM_CFM:
++ ssp_setup |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR);
++ break;
++ case SND_SOC_DAIFMT_CBM_CFS:
++ ssp_setup |= SSCR1_SCLKDIR;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFM:
++ ssp_setup |= SSCR1_SFRMDIR;
++ break;
++ }
++
++ switch(rtd->cpu_dai->dai_runtime.fmt) {
++ case SND_SOC_DAIFMT_CBS_CFS:
++ fmt = 1;
++ break;
++ case SND_SOC_DAIFMT_CBS_CFM:
++ fmt = 2;
++ break;
++ case SND_SOC_DAIFMT_CBM_CFS:
++ fmt = 3;
++ break;
++ }
++
++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].rx);
++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].tx);
++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].frm);
++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].clk);
++
++ switch (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
++ case SND_SOC_DAIFMT_NB_NF:
++ psp_mode |= SSPSP_SFRMP | SSPSP_FSRT;
++ break;
++ }
++
++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_DSP_A)
++ psp_mode |= SSPSP_SCMODE(2);
++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_DSP_B)
++ psp_mode |= SSPSP_SCMODE(3);
++
++ switch(clksrc[rtd->cpu_dai->id]) {
++ case 2: /* network clock */
++ ssp_mode |= SSCR0_NCS | SSCR0_MOD;
++ case 1: /* external clock */
++ ssp_mode |= SSCR0_ECS;
++ case 0: /* internal clock */
++ rate = get_scr(snd_soc_get_rate(rtd->cpu_dai->dai_runtime.pcmrate),
++ rtd->cpu_dai->id);
++ break;
++#ifdef CONFIG_PXA27x
++ case 3: /* audio clock */
++ ssp_mode |= (1 << 30);
++ SSACD_P(rtd->cpu_dai->id) = (0x1 << 3) |
++ pxa27x_set_audio_clk(
++ snd_soc_get_rate(rtd->cpu_dai->dai_runtime.pcmrate), fs);
++ break;
++#endif
++ }
++
++ ssp_config(&ssp[rtd->cpu_dai->id], ssp_mode, ssp_setup, psp_mode,
++ SSCR0_SerClkDiv(rate));
++#if 0
++ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n",
++ SSCR0_P(rtd->cpu_dai->id+1), SSCR1_P(rtd->cpu_dai->id+1),
++ SSTO_P(rtd->cpu_dai->id+1), SSPSP_P(rtd->cpu_dai->id+1),
++ SSSR_P(rtd->cpu_dai->id+1));
++#endif
++ return 0;
++}
++
++static int pxa2xx_ssp_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ int ret = 0;
++
++ switch (cmd) {
++ case SNDRV_PCM_TRIGGER_RESUME:
++ ssp_enable(&ssp[rtd->cpu_dai->id]);
++ break;
++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_TSRE;
++ else
++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_RSRE;
++ SSSR_P(rtd->cpu_dai->id+1) |= SSSR_P(rtd->cpu_dai->id+1);
++ break;
++ case SNDRV_PCM_TRIGGER_START:
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_TSRE;
++ else
++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_RSRE;
++ ssp_enable(&ssp[rtd->cpu_dai->id]);
++ break;
++ case SNDRV_PCM_TRIGGER_STOP:
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_TSRE;
++ else
++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_RSRE;
++ break;
++ case SNDRV_PCM_TRIGGER_SUSPEND:
++ ssp_disable(&ssp[rtd->cpu_dai->id]);
++ break;
++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_TSRE;
++ else
++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_RSRE;
++ break;
++
++ default:
++ ret = -EINVAL;
++ }
++#if 0
++ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n",
++ SSCR0_P(rtd->cpu_dai->id+1), SSCR1_P(rtd->cpu_dai->id+1),
++ SSTO_P(rtd->cpu_dai->id+1), SSPSP_P(rtd->cpu_dai->id+1),
++ SSSR_P(rtd->cpu_dai->id+1));
++#endif
++ return ret;
++}
++
++struct snd_soc_cpu_dai pxa_ssp_dai[] = {
++ { .name = "pxa2xx-ssp1",
++ .id = 0,
++ .type = SND_SOC_DAI_PCM,
++ .suspend = pxa2xx_ssp_suspend,
++ .resume = pxa2xx_ssp_resume,
++ .config_sysclk = pxa_ssp_config_sysclk,
++ .playback = {
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .channels_min = 1,
++ .channels_max = 2,},
++ .ops = {
++ .startup = pxa2xx_ssp_startup,
++ .shutdown = pxa2xx_ssp_shutdown,
++ .trigger = pxa2xx_ssp_trigger,
++ .hw_params = pxa2xx_ssp_hw_params,},
++ .caps = {
++ .mode = pxa2xx_ssp_modes,
++ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),},
++ },
++ { .name = "pxa2xx-ssp2",
++ .id = 1,
++ .type = SND_SOC_DAI_PCM,
++ .suspend = pxa2xx_ssp_suspend,
++ .resume = pxa2xx_ssp_resume,
++ .config_sysclk = pxa_ssp_config_sysclk,
++ .playback = {
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .channels_min = 1,
++ .channels_max = 2,},
++ .ops = {
++ .startup = pxa2xx_ssp_startup,
++ .shutdown = pxa2xx_ssp_shutdown,
++ .trigger = pxa2xx_ssp_trigger,
++ .hw_params = pxa2xx_ssp_hw_params,},
++ .caps = {
++ .mode = pxa2xx_ssp_modes,
++ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),},
++ },
++ { .name = "pxa2xx-ssp3",
++ .id = 2,
++ .type = SND_SOC_DAI_PCM,
++ .suspend = pxa2xx_ssp_suspend,
++ .resume = pxa2xx_ssp_resume,
++ .config_sysclk = pxa_ssp_config_sysclk,
++ .playback = {
++ .channels_min = 1,
++ .channels_max = 2,},
++ .capture = {
++ .channels_min = 1,
++ .channels_max = 2,},
++ .ops = {
++ .startup = pxa2xx_ssp_startup,
++ .shutdown = pxa2xx_ssp_shutdown,
++ .trigger = pxa2xx_ssp_trigger,
++ .hw_params = pxa2xx_ssp_hw_params,},
++ .caps = {
++ .mode = pxa2xx_ssp_modes,
++ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),},
++ },
++};
++
++EXPORT_SYMBOL_GPL(pxa_ssp_dai);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("pxa2xx SSP/PCM SoC Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/spitz.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/spitz.c
+@@ -0,0 +1,374 @@
++/*
++ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
++ * Richard Purdie <richard@openedhand.com>
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 30th Nov 2005 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/timer.h>
++#include <linux/interrupt.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/mach-types.h>
++#include <asm/hardware/scoop.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/hardware.h>
++#include <asm/arch/akita.h>
++#include <asm/arch/spitz.h>
++#include <asm/mach-types.h>
++#include "../codecs/wm8750.h"
++#include "pxa2xx-pcm.h"
++
++#define SPITZ_HP 0
++#define SPITZ_MIC 1
++#define SPITZ_LINE 2
++#define SPITZ_HEADSET 3
++#define SPITZ_HP_OFF 4
++#define SPITZ_SPK_ON 0
++#define SPITZ_SPK_OFF 1
++
++ /* audio clock in Hz - rounded from 12.235MHz */
++#define SPITZ_AUDIO_CLOCK 12288000
++
++static int spitz_jack_func;
++static int spitz_spk_func;
++
++static void spitz_ext_control(struct snd_soc_codec *codec)
++{
++ if (spitz_spk_func == SPITZ_SPK_ON)
++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
++ else
++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0);
++
++ /* set up jack connection */
++ switch (spitz_jack_func) {
++ case SPITZ_HP:
++ /* enable and unmute hp jack, disable mic bias */
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
++ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
++ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
++ break;
++ case SPITZ_MIC:
++ /* enable mic jack and bias, mute hp */
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
++ break;
++ case SPITZ_LINE:
++ /* enable line jack, disable mic bias and mute hp */
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 1);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
++ break;
++ case SPITZ_HEADSET:
++ /* enable and unmute headset jack enable mic bias, mute L hp */
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
++ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
++ break;
++ case SPITZ_HP_OFF:
++
++ /* jack removed, everything off */
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
++ break;
++ }
++ snd_soc_dapm_sync_endpoints(codec);
++}
++
++static int spitz_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec *codec = rtd->socdev->codec;
++
++ /* check the jack status at stream startup */
++ spitz_ext_control(codec);
++ return 0;
++}
++
++static struct snd_soc_ops spitz_ops = {
++ .startup = spitz_startup,
++};
++
++static int spitz_get_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = spitz_jack_func;
++ return 0;
++}
++
++static int spitz_set_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (spitz_jack_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ spitz_jack_func = ucontrol->value.integer.value[0];
++ spitz_ext_control(codec);
++ return 1;
++}
++
++static int spitz_get_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = spitz_spk_func;
++ return 0;
++}
++
++static int spitz_set_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (spitz_spk_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ spitz_spk_func = ucontrol->value.integer.value[0];
++ spitz_ext_control(codec);
++ return 1;
++}
++
++static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event)
++{
++ if (machine_is_borzoi() || machine_is_spitz()) {
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ set_scoop_gpio(&spitzscoop2_device.dev,
++ SPITZ_SCP2_MIC_BIAS);
++ else
++ reset_scoop_gpio(&spitzscoop2_device.dev,
++ SPITZ_SCP2_MIC_BIAS);
++ }
++
++ if (machine_is_akita()) {
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ akita_set_ioexp(&akitaioexp_device.dev,
++ AKITA_IOEXP_MIC_BIAS);
++ else
++ akita_reset_ioexp(&akitaioexp_device.dev,
++ AKITA_IOEXP_MIC_BIAS);
++ }
++ return 0;
++}
++
++/* spitz machine dapm widgets */
++static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
++ SND_SOC_DAPM_HP("Headphone Jack", NULL),
++ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
++ SND_SOC_DAPM_SPK("Ext Spk", NULL),
++ SND_SOC_DAPM_LINE("Line Jack", NULL),
++
++ /* headset is a mic and mono headphone */
++ SND_SOC_DAPM_HP("Headset Jack", NULL),
++};
++
++/* Spitz machine audio_map */
++static const char *audio_map[][3] = {
++
++ /* headphone connected to LOUT1, ROUT1 */
++ {"Headphone Jack", NULL, "LOUT1"},
++ {"Headphone Jack", NULL, "ROUT1"},
++
++ /* headset connected to ROUT1 and LINPUT1 with bias (def below) */
++ {"Headset Jack", NULL, "ROUT1"},
++
++ /* ext speaker connected to LOUT2, ROUT2 */
++ {"Ext Spk", NULL , "ROUT2"},
++ {"Ext Spk", NULL , "LOUT2"},
++
++ /* mic is connected to input 1 - with bias */
++ {"LINPUT1", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic Jack"},
++
++ /* line is connected to input 1 - no bias */
++ {"LINPUT1", NULL, "Line Jack"},
++
++ {NULL, NULL, NULL},
++};
++
++static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
++ "Off"};
++static const char *spk_function[] = {"On", "Off"};
++static const struct soc_enum spitz_enum[] = {
++ SOC_ENUM_SINGLE_EXT(5, jack_function),
++ SOC_ENUM_SINGLE_EXT(2, spk_function),
++};
++
++static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
++ SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
++ spitz_set_jack),
++ SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
++ spitz_set_spk),
++};
++
++/*
++ * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
++ */
++static int spitz_wm8750_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ /* NC codec pins */
++ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0);
++ snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0);
++ snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0);
++ snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0);
++ snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0);
++ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
++ snd_soc_dapm_set_endpoint(codec, "MONO", 0);
++
++ /* Add spitz specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
++ err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
++ if (err < 0)
++ return err;
++ }
++
++ /* Add spitz specific widgets */
++ for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
++ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
++ }
++
++ /* Set up spitz specific audio path audio_map */
++ for (i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++static unsigned int spitz_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++ struct snd_soc_clock_info *info)
++{
++ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) {
++ /* pxa2xx is i2s master */
++ switch (info->rate) {
++ case 11025:
++ case 22050:
++ case 44100:
++ case 88200:
++ /* configure codec digital filters
++ * for 11.025, 22.05, 44.1, 88.2 */
++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ 11289600);
++ break;
++ default:
++ /* configure codec digital filters for all other rates */
++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ SPITZ_AUDIO_CLOCK);
++ break;
++ }
++ /* configure pxa2xx i2s interface clocks as master */
++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info,
++ SPITZ_AUDIO_CLOCK);
++ } else {
++ /* codec is i2s master - only configure codec DAI clock */
++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info,
++ SPITZ_AUDIO_CLOCK);
++ }
++}
++
++/* spitz digital audio interface glue - connects codec <--> CPU */
++static struct snd_soc_dai_link spitz_dai = {
++ .name = "wm8750",
++ .stream_name = "WM8750",
++ .cpu_dai = &pxa_i2s_dai,
++ .codec_dai = &wm8750_dai,
++ .init = spitz_wm8750_init,
++ .config_sysclk = spitz_config_sysclk,
++};
++
++/* spitz audio machine driver */
++static struct snd_soc_machine snd_soc_machine_spitz = {
++ .name = "Spitz",
++ .dai_link = &spitz_dai,
++ .num_links = 1,
++ .ops = &spitz_ops,
++};
++
++/* spitz audio private data */
++static struct wm8750_setup_data spitz_wm8750_setup = {
++ .i2c_address = 0x1b,
++};
++
++/* spitz audio subsystem */
++static struct snd_soc_device spitz_snd_devdata = {
++ .machine = &snd_soc_machine_spitz,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8750,
++ .codec_data = &spitz_wm8750_setup,
++};
++
++static struct platform_device *spitz_snd_device;
++
++static int __init spitz_init(void)
++{
++ int ret;
++
++ if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
++ return -ENODEV;
++
++ spitz_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!spitz_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata);
++ spitz_snd_devdata.dev = &spitz_snd_device->dev;
++ ret = platform_device_add(spitz_snd_device);
++
++ if (ret)
++ platform_device_put(spitz_snd_device);
++
++ return ret;
++}
++
++static void __exit spitz_exit(void)
++{
++ platform_device_unregister(spitz_snd_device);
++}
++
++module_init(spitz_init);
++module_exit(spitz_exit);
++
++MODULE_AUTHOR("Richard Purdie");
++MODULE_DESCRIPTION("ALSA SoC Spitz");
++MODULE_LICENSE("GPL");
+Index: linux-2.6-pxa-new/sound/soc/pxa/tosa.c
+===================================================================
+--- /dev/null
++++ linux-2.6-pxa-new/sound/soc/pxa/tosa.c
+@@ -0,0 +1,287 @@
++/*
++ * tosa.c -- SoC audio for Tosa
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
++ * Richard Purdie <richard@openedhand.com>
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2 of the License, or (at your
++ * option) any later version.
++ *
++ * Revision history
++ * 30th Nov 2005 Initial version.
++ *
++ * GPIO's
++ * 1 - Jack Insertion
++ * 5 - Hookswitch (headset answer/hang up switch)
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.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 <asm/mach-types.h>
++#include <asm/hardware/tmio.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/hardware.h>
++#include <asm/arch/audio.h>
++#include <asm/arch/tosa.h>
++
++#include "../codecs/wm9712.h"
++#include "pxa2xx-pcm.h"
++
++static struct snd_soc_machine tosa;
++
++#define TOSA_HP 0
++#define TOSA_MIC_INT 1
++#define TOSA_HEADSET 2
++#define TOSA_HP_OFF 3
++#define TOSA_SPK_ON 0
++#define TOSA_SPK_OFF 1
++
++static int tosa_jack_func;
++static int tosa_spk_func;
++
++static void tosa_ext_control(struct snd_soc_codec *codec)
++{
++ int spk = 0, mic_int = 0, hp = 0, hs = 0;
++
++ /* set up jack connection */
++ switch (tosa_jack_func) {
++ case TOSA_HP:
++ hp = 1;
++ break;
++ case TOSA_MIC_INT:
++ mic_int = 1;
++ break;
++ case TOSA_HEADSET:
++ hs = 1;
++ break;
++ }
++
++ if (tosa_spk_func == TOSA_SPK_ON)
++ spk = 1;
++
++ snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
++ snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
++ snd_soc_dapm_sync_endpoints(codec);
++}
++
++static int tosa_startup(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec *codec = rtd->socdev->codec;
++
++ /* check the jack status at stream startup */
++ tosa_ext_control(codec);
++ return 0;
++}
++
++static struct snd_soc_ops tosa_ops = {
++ .startup = tosa_startup,
++};
++
++static int tosa_get_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = tosa_jack_func;
++ return 0;
++}
++
++static int tosa_set_jack(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (tosa_jack_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ tosa_jack_func = ucontrol->value.integer.value[0];
++ tosa_ext_control(codec);
++ return 1;
++}
++
++static int tosa_get_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ ucontrol->value.integer.value[0] = tosa_spk_func;
++ return 0;
++}
++
++static int tosa_set_spk(struct snd_kcontrol *kcontrol,
++ struct snd_ctl_elem_value *ucontrol)
++{
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++ if (tosa_spk_func == ucontrol->value.integer.value[0])
++ return 0;
++
++ tosa_spk_func = ucontrol->value.integer.value[0];
++ tosa_ext_control(codec);
++ return 1;
++}
++
++/* tosa dapm event handlers */
++static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event)
++{
++ if (SND_SOC_DAPM_EVENT_ON(event))
++ set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
++ else
++ reset_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
++ return 0;
++}
++
++/* tosa machine dapm widgets */
++static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
++SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
++SND_SOC_DAPM_HP("Headset Jack", NULL),
++SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
++SND_SOC_DAPM_SPK("Speaker", NULL),
++};
++
++/* tosa audio map */
++static const char *audio_map[][3] = {
++
++ /* headphone connected to HPOUTL, HPOUTR */
++ {"Headphone Jack", NULL, "HPOUTL"},
++ {"Headphone Jack", NULL, "HPOUTR"},
++
++ /* ext speaker connected to LOUT2, ROUT2 */
++ {"Speaker", NULL, "LOUT2"},
++ {"Speaker", NULL, "ROUT2"},
++
++ /* internal mic is connected to mic1, mic2 differential - with bias */
++ {"MIC1", NULL, "Mic Bias"},
++ {"MIC2", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic (Internal)"},
++
++ /* headset is connected to HPOUTR, and LINEINR with bias */
++ {"Headset Jack", NULL, "HPOUTR"},
++ {"LINEINR", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Headset Jack"},
++
++ {NULL, NULL, NULL},
++};
++
++static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
++ "Off"};
++static const char *spk_function[] = {"On", "Off"};
++static const struct soc_enum tosa_enum[] = {
++ SOC_ENUM_SINGLE_EXT(5, jack_function),
++ SOC_ENUM_SINGLE_EXT(2, spk_function),
++};
++