linux: Update 2.6.17 to latest asoc * untested in .dev *

2 files changed, 31716 insertions, 4 deletions

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