r598 - trunk/src/target/kernel/patches

laforge at sita.openmoko.org laforge at sita.openmoko.org
Wed Jan 24 12:45:08 CET 2007


Author: laforge
Date: 2007-01-24 12:45:04 +0100 (Wed, 24 Jan 2007)
New Revision: 598

Added:
   trunk/src/target/kernel/patches/asoc-v0.13rc1.patch
   trunk/src/target/kernel/patches/asoc-v013rc1-neo_backport.patch
   trunk/src/target/kernel/patches/gta01-vibrator.patch
   trunk/src/target/kernel/patches/s3c2410_udc.patch
Removed:
   trunk/src/target/kernel/patches/gta01_vibrator.patch
   trunk/src/target/kernel/patches/udc.patch
Modified:
   trunk/src/target/kernel/patches/gta01-s3c_mci-pdata.patch
   trunk/src/target/kernel/patches/gta01-vbus_draw.patch
   trunk/src/target/kernel/patches/qt2410-base.patch
   trunk/src/target/kernel/patches/qt2410-biglcd.patch
   trunk/src/target/kernel/patches/qt2410-s3c_mci-pdata.patch
   trunk/src/target/kernel/patches/qt2410-touchscreen.patch
   trunk/src/target/kernel/patches/s3c_mci_platform.patch
   trunk/src/target/kernel/patches/s3cmci-dma-free.patch
   trunk/src/target/kernel/patches/s3cmci_dbg.patch
   trunk/src/target/kernel/patches/series
Log:
- include ASoC v013.1 (plus non-working backport, rc2 pending today)
- use consisten patch naming
- reorder patches in logical (rather than chronological) order


Added: trunk/src/target/kernel/patches/asoc-v0.13rc1.patch
===================================================================
--- trunk/src/target/kernel/patches/asoc-v0.13rc1.patch	2007-01-24 09:35:25 UTC (rev 597)
+++ trunk/src/target/kernel/patches/asoc-v0.13rc1.patch	2007-01-24 11:45:04 UTC (rev 598)
@@ -0,0 +1,35486 @@
+Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/DAI.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/DAI.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,117 @@
++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
++======================
++
++DAI configuration parameters, all listed in include/sound/soc.h
++
++ 1) hardware DAI formats
++
++#define SND_SOC_DAIFMT_I2S        (1 << 0)	/* I2S mode */
++#define SND_SOC_DAIFMT_RIGHT_J    (1 << 1)	/* Right justified or LSB mode */
++#define SND_SOC_DAIFMT_LEFT_J     (1 << 2)	/* Left Justified or MSB 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 */
++
++#define SND_SOC_DAIFMT_MSB 	SND_SOC_DAIFMT_LEFT_J
++#define SND_SOC_DAIFMT_LSB	SND_SOC_DAIFMT_RIGHT_J
++
++ 2) Clock gating
++
++#define SND_SOC_DAIFMT_CONT			(0 << 4)	/* continuous clock */
++#define SND_SOC_DAIFMT_GATED		(1 << 4)	/* clock is gated when not Tx/Rx */
++
++ 3) 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 */
++
++ 4) 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 */
++
++
++ASoC DAI Operations
++===================
++
++	/* codec DAI clocking configuration */
++	int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir);
++	int (*set_pll)(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out);
++	int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div);
++
++	/* CPU DAI format configuration */
++	int (*set_fmt)(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt);
++	int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai,
++		unsigned int mask, int slots);
++	int (*set_tristate)(struct snd_soc_codec_dai *, int tristate);
++
++
+Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/clocking.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/clocking.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,47 @@
++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. In general
++it's best to configure BCLK to the lowest possible speed (depending on your
++rate, number of channels and wordsize) to save on power.
++
++
+Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/codec.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/codec.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,196 @@
++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) 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.
++
++
++2 - 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,
++	},
++	.ops = {
++		.prepare = wm8731_pcm_prepare,
++	},
++};
++
++
++3 - 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;
++
++
++4 - 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
++
++5 - 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
++
++
++6 - 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.
++
++
++7 - 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 */
++
++
++8 - 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-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/dapm.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/dapm.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/machine.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/machine.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,113 @@
++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,
++	.ops = &corgi_ops,
++};
++
++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,
++};
++
++
++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-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/overview.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/overview.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/platform.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/platform.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/pops_clicks.txt
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/pops_clicks.txt	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/include/sound/ac97_codec.h
+===================================================================
+--- linux-2.6.17.14-fic4.test.orig/include/sound/ac97_codec.h	2006-10-13 20:55:04.000000000 +0200
++++ linux-2.6.17.14-fic4.test/include/sound/ac97_codec.h	2007-01-24 12:19:24.000000000 +0100
+@@ -407,6 +407,7 @@
+ 
+ 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-2.6.17.14-fic4.test/include/sound/soc-dapm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/include/sound/soc-dapm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/include/sound/soc.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/include/sound/soc.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,460 @@
++/*
++ * 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 "0.13.0rc1"
++
++/*
++ * 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, xshift, xmask, xinvert,\
++	 xhandler_get, xhandler_put) \
++{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
++	.info = snd_soc_info_volsw, \
++	.get = xhandler_get, .put = xhandler_put, \
++	.private_value = SOC_SINGLE_VALUE(xreg, xshift, 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		0	/* I2S mode */
++#define SND_SOC_DAIFMT_RIGHT_J	1	/* Right justified mode */
++#define SND_SOC_DAIFMT_LEFT_J	2	/* Left Justified mode */
++#define SND_SOC_DAIFMT_DSP_A	3	/* L data msb after FRM or LRC */
++#define SND_SOC_DAIFMT_DSP_B	4	/* L data msb during FRM or LRC */
++#define SND_SOC_DAIFMT_AC97		5	/* AC97 */
++
++#define SND_SOC_DAIFMT_MSB 	SND_SOC_DAIFMT_LEFT_J
++#define SND_SOC_DAIFMT_LSB	SND_SOC_DAIFMT_RIGHT_J
++
++/*
++ * DAI Gating
++ */
++#define SND_SOC_DAIFMT_CONT			(0 << 4)	/* continuous clock */
++#define SND_SOC_DAIFMT_GATED		(1 << 4)	/* clock is gated when not Tx/Rx */
++
++/*
++ * DAI hardware signal inversions
++ */
++#define SND_SOC_DAIFMT_NB_NF		(0 << 8)	/* normal bit clock + frame */
++#define SND_SOC_DAIFMT_NB_IF		(1 << 8)	/* normal bclk + inv frm */
++#define SND_SOC_DAIFMT_IB_NF		(2 << 8)	/* invert bclk + nor frm */
++#define SND_SOC_DAIFMT_IB_IF		(3 << 8)	/* 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	(0 << 12) /* codec clk & frm master */
++#define SND_SOC_DAIFMT_CBS_CFM	(1 << 12) /* codec clk slave & frm master */
++#define SND_SOC_DAIFMT_CBM_CFS	(2 << 12) /* codec clk master & frame slave */
++#define SND_SOC_DAIFMT_CBS_CFS	(3 << 12) /* codec clk & frm slave */
++
++#define SND_SOC_DAIFMT_FORMAT_MASK		0x000f
++#define SND_SOC_DAIFMT_CLOCK_MASK		0x00f0
++#define SND_SOC_DAIFMT_INV_MASK			0x0f00
++#define SND_SOC_DAIFMT_MASTER_MASK		0xf000
++
++
++/*
++ * Master Clock Directions
++ */
++#define SND_SOC_CLOCK_IN	0
++#define SND_SOC_CLOCK_OUT	1
++
++/*
++ * 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;
++	u64 formats;			/* SNDRV_PCM_FMTBIT_* */
++	unsigned int rates;		/* SNDRV_PCM_RATE_* */
++	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 */
++};
++
++/* ASoC alsa 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);
++};
++
++/* ASoC codec DAI ops */
++struct snd_soc_codec_ops {
++	/* codec DAI clocking configuration */
++	int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir);
++	int (*set_pll)(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out);
++	int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div);
++
++	/* CPU DAI format configuration */
++	int (*set_fmt)(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt);
++	int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai,
++		unsigned int mask, int slots);
++	int (*set_tristate)(struct snd_soc_codec_dai *, int tristate);
++
++	/* digital mute */
++	int (*digital_mute)(struct snd_soc_codec_dai *, int mute);
++};
++
++/* ASoC cpu DAI ops */
++struct snd_soc_cpu_ops {
++	/* CPU DAI clocking configuration */
++	int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai,
++		int clk_id, unsigned int freq, int dir);
++	int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai,
++		int div_id, int div);
++	int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out);
++
++	/* CPU DAI format configuration */
++	int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int fmt);
++	int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int mask, int slots);
++	int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate);
++};
++
++/* 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;
++
++	/* DAI runtime info */
++	struct snd_soc_codec *codec;
++	unsigned int active;
++	unsigned char pop_wait:1;
++
++	/* ops */
++	struct snd_soc_ops ops;
++	struct snd_soc_codec_ops dai_ops;
++
++	/* DAI private data */
++	void *private_data;
++};
++
++/* SoC CPU DAI */
++struct snd_soc_cpu_dai {
++
++	/* DAI description */
++	char *name;
++	unsigned int id;
++	unsigned char type;
++
++	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);
++
++	/* ops */
++	struct snd_soc_ops ops;
++	struct snd_soc_cpu_ops dai_ops;
++
++	/* DAI capabilities */
++	struct snd_soc_pcm_stream capture;
++	struct snd_soc_pcm_stream playback;
++
++	/* DAI runtime info */
++	struct snd_pcm_runtime *runtime;
++	unsigned char active:1;
++	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;
++	struct delayed_work delayed_work;
++	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;
++
++	/* machine stream operations */
++	struct snd_soc_ops *ops;
++
++	/* codec/machine specific init - e.g. add machine controls */
++	int (*init)(struct snd_soc_codec *codec);
++};
++
++/* 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);
++
++	/* 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;
++	struct delayed_work close_work;
++};
++
++/* runtime channel data */
++struct snd_soc_pcm_runtime {
++	struct snd_soc_dai_link *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;
++};
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/Kconfig
+===================================================================
+--- linux-2.6.17.14-fic4.test.orig/sound/Kconfig	2006-10-13 20:55:04.000000000 +0200
++++ linux-2.6.17.14-fic4.test/sound/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -74,6 +74,8 @@
+ 
+ source "sound/parisc/Kconfig"
+ 
++source "sound/soc/Kconfig"
++
+ endmenu
+ 
+ menu "Open Sound System"
+Index: linux-2.6.17.14-fic4.test/sound/soc/Kconfig
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/sound/soc/Makefile
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/Makefile	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/sound/soc/codecs/Kconfig
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,102 @@
++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_WM8956
++	tristate "SoC driver for the WM8956 codec"
++	depends SND_SOC
++	help
++	  Say Y or M if you want to support the WM8956 codec.
++
++config SND_SOC_WM8960
++	tristate "SoC driver for the WM8960 codec"
++	depends SND_SOC
++	help
++	  Say Y or M if you want to support the WM8960 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-2.6.17.14-fic4.test/sound/soc/codecs/Makefile
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/Makefile	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,35 @@
++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-wm8956-objs := wm8956.o
++snd-soc-wm8960-objs := wm8960.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_WM8956)	+= snd-soc-wm8956.o
++obj-$(CONFIG_SND_SOC_WM8960)	+= snd-soc-wm8960.o
++obj-$(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-2.6.17.14-fic4.test/sound/soc/codecs/ac97.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,156 @@
++/*
++ * ac97.c  --  ALSA Soc AC97 codec support
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  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 "0.6"
++
++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);
++}
++
++#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
++
++static struct snd_soc_codec_dai ac97_dai = {
++	.name = "AC97 HiFi",
++	.playback = {
++		.stream_name = "AC97 Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = STD_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "AC97 Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = STD_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.prepare = ac97_prepare,},
++};
++
++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-2.6.17.14-fic4.test/sound/soc/codecs/ac97.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.h	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,690 @@
++/*
++ * ak4535.c  --  AK4535 ALSA Soc Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard at openedhand.com>
++ *
++ * Based on wm8753.c by Liam Girdwood
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "ak4535.h"
++
++#define AUDIO_NAME "ak4535"
++#define AK4535_VERSION "0.3"
++
++struct snd_soc_codec_device soc_codec_dev_ak4535;
++
++/* codec private data */
++struct ak4535_priv {
++	unsigned int sysclk;
++};
++
++/*
++ * ak4535 register cache
++ */
++static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
++    0x0000, 0x0080, 0x0000, 0x0003,
++    0x0002, 0x0000, 0x0011, 0x0001,
++    0x0000, 0x0040, 0x0036, 0x0010,
++    0x0000, 0x0000, 0x0057, 0x0000,
++};
++
++/*
++ * read ak4535 register cache
++ */
++static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= AK4535_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write ak4535 register cache
++ */
++static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= AK4535_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the AK4535 register space
++ */
++static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D8 AK4535 register offset
++	 *   D7...D0 register data
++	 */
++	data[0] = reg & 0xff;
++	data[1] = value & 0xff;
++
++	ak4535_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -EIO;
++}
++
++static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
++static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
++static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
++static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"};
++static const char *ak4535_mic_select[] = {"Internal", "External"};
++
++static const struct soc_enum ak4535_enum[] = {
++	SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain),
++	SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out),
++	SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out),
++	SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp),
++	SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select),
++};
++
++static const struct snd_kcontrol_new ak4535_snd_controls[] = {
++	SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0),
++	SOC_ENUM("Mono 1 Output", ak4535_enum[1]),
++	SOC_ENUM("Mono 1 Gain", ak4535_enum[0]),
++	SOC_ENUM("Headphone Output", ak4535_enum[2]),
++	SOC_ENUM("Playback Deemphasis", ak4535_enum[3]),
++	SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0),
++	SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0),
++	SOC_ENUM("Mic Select", ak4535_enum[4]),
++	SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0),
++	SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0),
++	SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0),
++	SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0),
++	SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0),
++	SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0),
++	SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0),
++	SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1),
++	SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1),
++	SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0),
++	SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
++};
++
++/* add non dapm controls */
++static int ak4535_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++			snd_soc_cnew(&ak4535_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* Mono 1 Mixer */
++static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
++	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
++	SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0),
++};
++
++/* Stereo Mixer */
++static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = {
++	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0),
++	SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0),
++	SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0),
++};
++
++/* Input Mixer */
++static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = {
++	SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0),
++	SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0),
++};
++
++/* Input mux */
++static const struct snd_kcontrol_new ak4535_input_mux_control =
++	SOC_DAPM_ENUM("Input Select", ak4535_enum[0]);
++
++/* HP L switch */
++static const struct snd_kcontrol_new ak4535_hpl_control =
++	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1);
++
++/* HP R switch */
++static const struct snd_kcontrol_new ak4535_hpr_control =
++	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1);
++
++/* Speaker switch */
++static const struct snd_kcontrol_new ak4535_spk_control =
++	SOC_DAPM_SINGLE("Switch", AK4535_MODE2, 0, 0, 0);
++
++/* mono 2 switch */
++static const struct snd_kcontrol_new ak4535_mono2_control =
++	SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0);
++
++/* Line out switch */
++static const struct snd_kcontrol_new ak4535_line_control =
++	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0);
++
++/* ak4535 dapm widgets */
++static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = {
++	SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
++		&ak4535_stereo_mixer_controls[0],
++		ARRAY_SIZE(ak4535_stereo_mixer_controls)),
++	SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
++		&ak4535_mono1_mixer_controls[0],
++		ARRAY_SIZE(ak4535_mono1_mixer_controls)),
++	SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
++		&ak4535_input_mixer_controls[0],
++		ARRAY_SIZE(ak4535_mono1_mixer_controls)),
++	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
++		&ak4535_input_mux_control),
++	SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0),
++	SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
++		&ak4535_mono2_control),
++	SND_SOC_DAPM_SWITCH("Speaker Enable", SND_SOC_NOPM, 0, 0,
++		&ak4535_spk_control),
++	SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
++		&ak4535_line_control),
++	SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0,
++		&ak4535_hpl_control),
++	SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0,
++		&ak4535_hpr_control),
++	SND_SOC_DAPM_OUTPUT("LOUT"),
++	SND_SOC_DAPM_OUTPUT("HPL"),
++	SND_SOC_DAPM_OUTPUT("ROUT"),
++	SND_SOC_DAPM_OUTPUT("HPR"),
++	SND_SOC_DAPM_OUTPUT("SPP"),
++	SND_SOC_DAPM_OUTPUT("SPN"),
++	SND_SOC_DAPM_OUTPUT("MOUT1"),
++	SND_SOC_DAPM_OUTPUT("MOUT2"),
++	SND_SOC_DAPM_OUTPUT("MICOUT"),
++	SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 1),
++	SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0),
++
++	SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0),
++	SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0),
++	SND_SOC_DAPM_INPUT("MICIN"),
++	SND_SOC_DAPM_INPUT("MICEXT"),
++	SND_SOC_DAPM_INPUT("AUX"),
++	SND_SOC_DAPM_INPUT("MIN"),
++	SND_SOC_DAPM_INPUT("AIN"),
++};
++
++static const char *audio_map[][3] = {
++	/*stereo mixer */
++	{"Stereo Mixer", "Playback Switch", "DAC"},
++	{"Stereo Mixer", "Mic Sidetone Switch", "Mic"},
++	{"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
++
++	/* mono1 mixer */
++	{"Mono1 Mixer", "Mic Sidetone Switch", "Mic"},
++	{"Mono1 Mixer", "Mono Playback Switch", "DAC"},
++
++	/* mono2 mixer */
++	{"Mono2 Mixer", "Mono Playback Switch", "Stereo Mixer"},
++
++	/* Mic */
++	{"AIN", NULL, "Mic"},
++	{"Input Mux", "Internal", "Mic Int Bias"},
++	{"Input Mux", "External", "Mic Ext Bias"},
++	{"Mic Int Bias", NULL, "MICIN"},
++	{"Mic Ext Bias", NULL, "MICEXT"},
++	{"MICOUT", NULL, "Input Mux"},
++
++	/* line out */
++	{"LOUT", "Switch", "Line"},
++	{"ROUT", "Switch", "Line Out Enable"},
++	{"Line Out Enable", NULL, "Line Out"},
++	{"Line Out", NULL, "Stereo Mixer"},
++
++	/* mono1 out */
++	{"MOUT1", NULL, "Mono Out"},
++	{"Mono Out", NULL, "Mono Mixer"},
++
++	/* left HP */
++	{"HPL", "Switch", "Left HP Enable"},
++	{"Left HP Enable", NULL, "HP L Amp"},
++	{"HP L Amp", NULL, "Stereo Mixer"},
++
++	/* right HP */
++	{"HPR", "Switch", "Right HP Enable"},
++	{"Right HP Enable", NULL, "HP R Amp"},
++	{"HP R Amp", NULL, "Stereo Mixer"},
++
++	/* speaker */
++	{"SPP", "Switch", "Speaker Enable"},
++	{"SPN", "Switch", "Speaker Enable"},
++	{"Speaker Enable", NULL, "Spk Amp"},
++	{"Spk Amp", NULL, "MIN"},
++
++	/* mono 2 */
++	{"MOUT2", "Switch", "Mono 2 Enable"},
++	{"Mono 2 Enable", NULL, "Stereo Mixer"},
++
++	/* Aux In */
++	{"Aux In", NULL, "AUX"},
++
++	/* ADC */
++	{"ADC", NULL, "Input Mixer"},
++	{"Input Mixer", "Mic Capture Switch", "Mic"},
++	{"Input Mixer", "Aux Capture Switch", "Aux In"},
++
++	/* terminator */
++	{NULL, NULL, NULL},
++};
++
++static int ak4535_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(ak4535_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &ak4535_dapm_widgets[i]);
++	}
++
++	/* set up audio path audio_mapnects */
++	for(i = 0; audio_map[i][0] != NULL; i++) {
++		snd_soc_dapm_connect_input(codec, audio_map[i][0],
++			audio_map[i][1], audio_map[i][2]);
++	}
++
++	snd_soc_dapm_new_widgets(codec);
++	return 0;
++}
++
++static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++	int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct ak4535_priv *ak4535 = codec->private_data;
++
++	ak4535->sysclk = freq;
++	return 0;
++}
++
++static int ak4535_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct ak4535_priv *ak4535 = codec->private_data;
++	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
++	int rate = params_rate(params), fs = 256;
++
++	if (rate)
++		fs = ak4535->sysclk / rate;
++
++	/* set fs */
++	switch (fs) {
++	case 1024:
++		mode2 |= (0x2 << 5);
++		break;
++	case 512:
++		mode2 |= (0x1 << 5);
++		break;
++	case 256:
++		break;
++	}
++
++	/* set rate */
++	ak4535_write(codec, AK4535_MODE2, mode2);
++	return 0;
++}
++
++static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u8 mode1 = 0;
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		mode1 = 0x0002;
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		mode1 = 0x0001;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* use 32 fs for BCLK to save power */
++	mode1 |= 0x4;
++
++	ak4535_write(codec, AK4535_MODE1, mode1);
++	return 0;
++}
++
++static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
++	if (mute)
++		ak4535_write(codec, AK4535_DAC, mute_reg);
++	else
++		ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
++	return 0;
++}
++
++static int ak4535_dapm_event(struct snd_soc_codec *codec, int event)
++{
++	switch (event) {
++	case SNDRV_CTL_POWER_D0: /* full On */
++	/* vref/mid, clk and osc on, dac unmute, active */
++	case SNDRV_CTL_POWER_D1: /* partial On */
++	case SNDRV_CTL_POWER_D2: /* partial On */
++		break;
++	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++		/* everything off except vref/vmid, dac mute, inactive */
++		ak4535_write(codec, AK4535_PM1, 0x80);
++		ak4535_write(codec, AK4535_PM2, 0x0);
++		break;
++	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++		/* everything off, inactive */
++		ak4535_write(codec, AK4535_PM1, 0x0);
++		ak4535_write(codec, AK4535_PM2, 0x80);
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000)
++
++struct snd_soc_codec_dai ak4535_dai = {
++	.name = "AK4535",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = AK4535_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = AK4535_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = ak4535_hw_params,
++	},
++	.dai_ops = {
++		.set_fmt = ak4535_set_dai_fmt,
++		.digital_mute = ak4535_mute,
++		.set_sysclk = ak4535_set_dai_sysclk,
++	},
++};
++EXPORT_SYMBOL_GPL(ak4535_dai);
++
++static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++
++	ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int ak4535_resume(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++	int i;
++	u8 data[2];
++	u16 *cache = codec->reg_cache;
++
++	/* Sync reg_cache with the hardware */
++	for (i = 0; i < ARRAY_SIZE(ak4535_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	ak4535_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the AK4535 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int ak4535_init(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int ret = 0;
++
++	codec->name = "AK4535";
++	codec->owner = THIS_MODULE;
++	codec->read = ak4535_read_reg_cache;
++	codec->write = ak4535_write;
++	codec->dapm_event = ak4535_dapm_event;
++	codec->dai = &ak4535_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(ak4535_reg);
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(ak4535_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, ak4535_reg,
++		sizeof(u16) * ARRAY_SIZE(ak4535_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ak4535_reg);
++
++	/* register pcms */
++	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++	if (ret < 0) {
++		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;
++	struct ak4535_priv *ak4535;
++	int ret = 0;
++
++	printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
++
++	setup = socdev->codec_data;
++	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++	if (codec == NULL)
++		return -ENOMEM;
++
++	ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL);
++	if (ak4535 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = ak4535;
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++
++	ak4535_socdev = socdev;
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++	if (setup->i2c_address) {
++		normal_i2c[0] = setup->i2c_address;
++		codec->hw_write = (hw_write_t)i2c_master_send;
++		ret = i2c_add_driver(&ak4535_i2c_driver);
++		if (ret != 0)
++			printk(KERN_ERR "can't add i2c driver");
++	}
++#else
++	/* Add other interfaces here */
++#endif
++	return ret;
++}
++
++/* power down chip */
++static int ak4535_remove(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec* codec = socdev->codec;
++
++	if (codec->control_data)
++		ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++	snd_soc_free_pcms(socdev);
++	snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++	i2c_del_driver(&ak4535_i2c_driver);
++#endif
++	kfree(codec->private_data);
++	kfree(codec);
++
++	return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_ak4535 = {
++	.probe = 	ak4535_probe,
++	.remove = 	ak4535_remove,
++	.suspend = 	ak4535_suspend,
++	.resume =	ak4535_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
++
++MODULE_DESCRIPTION("Soc AK4535 driver");
++MODULE_AUTHOR("Richard Purdie");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,46 @@
++/*
++ * ak4535.h  --  AK4535 Soc Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard at openedhand.com>
++ *
++ * Based on wm8753.h
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef _AK4535_H
++#define _AK4535_H
++
++/* AK4535 register space */
++
++#define AK4535_PM1		0x0
++#define AK4535_PM2		0x1
++#define AK4535_SIG1		0x2
++#define AK4535_SIG2		0x3
++#define AK4535_MODE1	0x4
++#define AK4535_MODE2	0x5
++#define AK4535_DAC		0x6
++#define AK4535_MIC		0x7
++#define AK4535_TIMER	0x8
++#define AK4535_ALC1		0x9
++#define AK4535_ALC2		0xa
++#define AK4535_PGA		0xb
++#define AK4535_LATT		0xc
++#define AK4535_RATT		0xd
++#define AK4535_VOL		0xe
++#define AK4535_STATUS	0xf
++
++#define AK4535_CACHEREGNUM 	0x10
++
++struct ak4535_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai ak4535_dai;
++extern struct snd_soc_codec_device soc_codec_dev_ak4535;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,728 @@
++/*
++ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ * Modified by Richard Purdie <richard at openedhand.com> to fit into SoC
++ * codec model.
++ *
++ * Copyright (c) 2005 Giorgio Padrin <giorgio at mandarinlogiq.org>
++ * Copyright 2005 Openedhand Ltd.
++ */
++
++#include <linux/module.h>
++#include <linux/init.h>
++#include <linux/types.h>
++#include <linux/string.h>
++#include <linux/slab.h>
++#include <linux/errno.h>
++#include <linux/ioctl.h>
++#include <linux/delay.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/control.h>
++#include <sound/initval.h>
++#include <sound/info.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include "uda1380.h"
++
++#define UDA1380_VERSION "0.5"
++
++/*
++ * uda1380 register cache
++ */
++static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
++    0x0502, 0x0000, 0x0000, 0x3f3f,
++    0x0202, 0x0000, 0x0000, 0x0000,
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0000, 0xff00, 0x0000, 0x4800,
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0000, 0x8000, 0x0002, 0x0000,
++};
++
++/*
++ * read uda1380 register cache
++ */
++static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg == UDA1380_RESET)
++		return 0;
++	if (reg >= UDA1380_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write uda1380 register cache
++ */
++static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= UDA1380_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the UDA1380 register space
++ */
++static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[3];
++
++	/* data is
++	 *   data[0] is register offset
++	 *   data[1] is MS byte
++	 *   data[2] is LS byte
++	 */
++	data[0] = reg;
++	data[1] = (value & 0xff00) >> 8;
++	data[2] = value & 0x00ff;
++
++	uda1380_write_reg_cache (codec, reg, value);
++
++	/* the interpolator & decimator regs must only be written when the
++	 * codec DAI is active.
++	 */
++	if (!codec->active && (reg >= UDA1380_MVOL))
++		return 0;
++printk("uda hw write %x val %x\n", reg, value);
++	if (codec->hw_write(codec->control_data, data, 3) == 3) {
++		unsigned int val;
++		i2c_master_send(codec->control_data, data, 1);
++		i2c_master_recv(codec->control_data, data, 2);
++		val = (data[0]<<8) | data[1];
++		if (val != value) {
++			printk("READ BACK VAL %x\n", (data[0]<<8) | data[1]);
++			return -EIO;
++		}
++		return 0;
++	} else
++		return -EIO;
++}
++
++#define uda1380_reset(c)	uda1380_write(c, UDA1380_RESET, 0)
++
++/* declarations of ALSA reg_elem_REAL controls */
++static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz",
++	"96kHz"};
++static const char *uda1380_input_sel[] = {"Line", "Mic"};
++static const char *uda1380_output_sel[] = {"Direct", "Mixer"};
++static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2", "Maximum"};
++
++static const struct soc_enum uda1380_enum[] = {
++	SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp),
++	SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel),
++	SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode),
++	SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel), /* R02_EN_AVC */
++};
++
++static const struct snd_kcontrol_new uda1380_snd_controls[] = {
++	SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 255, 1),
++	SOC_DOUBLE("Mixer Volume", UDA1380_MIXVOL, 0, 8, 255, 1),
++	SOC_ENUM("Sound Processing Filter Mode", uda1380_enum[2]),
++	SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0),
++	SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0),
++	SOC_ENUM("Playback De-emphasis", uda1380_enum[0]),
++	SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0),
++	SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0),
++	SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0),
++	SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 1),
++	SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0),
++	SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0),
++	SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1),
++	SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
++	SOC_SINGLE("Silence", UDA1380_MIXER, 7, 1, 0),
++	SOC_SINGLE("Silence Detection", UDA1380_MIXER, 6, 1, 0),
++};
++
++/* add non dapm controls */
++static int uda1380_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++			snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* Input mux */
++static const struct snd_kcontrol_new uda1380_input_mux_control =
++	SOC_DAPM_ENUM("Input Select", uda1380_enum[1]);
++
++/* Output mux */
++static const struct snd_kcontrol_new uda1380_output_mux_control =
++	SOC_DAPM_ENUM("Output Select", uda1380_enum[3]);
++
++static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
++	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
++		&uda1380_input_mux_control),
++	SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
++		&uda1380_output_mux_control),
++	SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
++	SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
++	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
++	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
++	SND_SOC_DAPM_INPUT("VINM"),
++	SND_SOC_DAPM_INPUT("VINL"),
++	SND_SOC_DAPM_INPUT("VINR"),
++	SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
++	SND_SOC_DAPM_OUTPUT("VOUTLHP"),
++	SND_SOC_DAPM_OUTPUT("VOUTRHP"),
++	SND_SOC_DAPM_OUTPUT("VOUTL"),
++	SND_SOC_DAPM_OUTPUT("VOUTR"),
++	SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
++	SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
++};
++
++static const char *audio_map[][3] = {
++
++	/* output mux */
++	{"HeadPhone Driver", NULL, "Output Mux"},
++	{"VOUTR", NULL, "Output Mux"},
++	{"VOUTL", NULL, "Output Mux"},
++
++	{"Analog Mixer", NULL, "VINR"},
++	{"Analog Mixer", NULL, "VINL"},
++	{"Analog Mixer", NULL, "DAC"},
++
++	{"Output Mux", "Direct", "DAC"},
++	{"Output Mux", "Mixer", "Analog Mixer"},
++
++	/* headphone driver */
++	{"VOUTLHP", NULL, "HeadPhone Driver"},
++	{"VOUTRHP", NULL, "HeadPhone Driver"},
++
++	/* input mux */
++	{"Left ADC", NULL, "Input Mux"},
++	{"Input Mux", "Mic", "Mic LNA"},
++	{"Input Mux", "Line", "Left PGA"},
++
++	/* right input */
++	{"Right ADC", NULL, "Right PGA"},
++
++	/* inputs */
++	{"Mic LNA", NULL, "VINM"},
++	{"Left PGA", NULL, "VINL"},
++	{"Right PGA", NULL, "VINR"},
++
++	/* terminator */
++	{NULL, NULL, NULL},
++};
++
++static int uda1380_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
++	}
++
++	/* set up audio path interconnects */
++	for(i = 0; audio_map[i][0] != NULL; i++) {
++		snd_soc_dapm_connect_input(codec, audio_map[i][0],
++			audio_map[i][1], audio_map[i][2]);
++	}
++
++	snd_soc_dapm_new_widgets(codec);
++	return 0;
++}
++
++static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	int iface;
++	/* set up DAI based upon fmt */
++
++	iface = uda1380_read_reg_cache (codec, UDA1380_IFACE);
++	iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
++
++	/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
++	switch(fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		iface |= R01_SFORI_I2S | R01_SFORO_I2S;
++		break;
++	case SND_SOC_DAIFMT_LSB:
++		iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
++		break;
++	case SND_SOC_DAIFMT_MSB:
++		iface |= R01_SFORI_MSB | R01_SFORO_I2S;
++	}
++
++	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) {
++		iface |= R01_SIM;
++	}
++
++	uda1380_write(codec, UDA1380_IFACE, iface);
++
++	return 0;
++}
++
++/*
++ * Flush reg cache
++ * We can only write the interpolator and decimator registers
++ * when the DAI is being clocked by the CPU DAI. It's up to the
++ * machine and cpu DAI driver to do this before we are called.
++ */
++static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	int reg, reg_start, reg_end, clk;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		reg_start = UDA1380_MVOL;
++		reg_end = UDA1380_MIXER;
++	} else {
++		reg_start = UDA1380_DEC;
++		reg_end = UDA1380_AGC;
++	}
++
++	/* FIXME disable DAC_CLK */
++	clk = uda1380_read_reg_cache (codec, 00);
++	uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
++
++	for (reg = reg_start; reg <= reg_end; reg++ ) {
++		printk("reg %x val %x\n",reg, uda1380_read_reg_cache (codec, reg));
++		uda1380_write(codec, reg, uda1380_read_reg_cache (codec, reg));
++	}
++
++	/* FIXME enable DAC_CLK */
++	uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
++
++	return 0;
++}
++
++static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
++
++	/* set WSPLL power and divider if running from this clock */
++	if (clk & R00_DAC_CLK) {
++		int rate = params_rate(params);
++		u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
++		clk &= ~0x3; /* clear SEL_LOOP_DIV */
++		switch (rate) {
++		case 6250 ... 12500:
++			clk |= 0x0;
++			break;
++		case 12501 ... 25000:
++			clk |= 0x1;
++			break;
++		case 25001 ... 50000:
++			clk |= 0x2;
++			break;
++		case 50001 ... 100000:
++			clk |= 0x3;
++			break;
++		}
++		uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
++	}
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		clk |= R00_EN_DAC | R00_EN_INT;
++	else
++		clk |= R00_EN_ADC | R00_EN_DEC;
++
++	uda1380_write(codec, UDA1380_CLK, clk);
++	return 0;
++}
++
++static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
++
++	/* shut down WSPLL power if running from this clock */
++	if (clk & R00_DAC_CLK) {
++		u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
++		uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
++	}
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		clk &= ~(R00_EN_DAC | R00_EN_INT);
++	else
++		clk &= ~(R00_EN_ADC | R00_EN_DEC);
++
++	uda1380_write(codec, UDA1380_CLK, clk);
++}
++
++static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff;
++
++	/* FIXME: mute(codec,0) is called when the magician clock is already
++	 * set to WSPLL, but for some unknown reason writing to interpolator
++	 * registers works only when clocked by SYSCLK */
++	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
++	uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
++	if(mute)
++		uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000);
++	else
++		uda1380_write(codec, UDA1380_DEEMP, mute_reg);
++	uda1380_write(codec, UDA1380_CLK, clk);
++	return 0;
++}
++
++static int uda1380_dapm_event(struct snd_soc_codec *codec, int event)
++{
++	int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
++
++	switch (event) {
++		case SNDRV_CTL_POWER_D0: /* full On */
++		case SNDRV_CTL_POWER_D1: /* partial On */
++		case SNDRV_CTL_POWER_D2: /* partial On */
++			/* enable internal bias */
++			uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
++			break;
++		case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++			/* everything off except internal bias */
++			uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
++			break;
++		case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++			/* everything off, inactive */
++			uda1380_write(codec, UDA1380_PM, 0x0);
++			break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000)
++
++struct snd_soc_codec_dai uda1380_dai[] = {
++{
++	.name = "UDA1380",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = UDA1380_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = UDA1380_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = uda1380_pcm_hw_params,
++		.shutdown = uda1380_pcm_shutdown,
++		.prepare = uda1380_pcm_prepare,
++	},
++	.dai_ops = {
++		.digital_mute = uda1380_mute,
++		.set_fmt = uda1380_set_dai_fmt,
++	},
++},
++{/* playback only  - dual interface */
++	.name = "UDA1380",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = UDA1380_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,
++	},
++	.ops = {
++		.hw_params = uda1380_pcm_hw_params,
++		.shutdown = uda1380_pcm_shutdown,
++		.prepare = uda1380_pcm_prepare,
++	},
++	.dai_ops = {
++		.digital_mute = uda1380_mute,
++		.set_fmt = uda1380_set_dai_fmt,
++	},
++},
++{ /* capture only - dual interface*/
++	.name = "UDA1380",
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = UDA1380_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,
++	},
++	.ops = {
++		.hw_params = uda1380_pcm_hw_params,
++		.shutdown = uda1380_pcm_shutdown,
++		.prepare = uda1380_pcm_prepare,
++	},
++	.dai_ops = {
++		.set_fmt = uda1380_set_dai_fmt,
++	},
++},
++};
++EXPORT_SYMBOL_GPL(uda1380_dai);
++
++static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++
++	uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int uda1380_resume(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++	int i;
++	u8 data[2];
++	u16 *cache = codec->reg_cache;
++
++	/* Sync reg_cache with the hardware */
++	for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	uda1380_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the UDA1380 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int ret = 0;
++printk("uda1380 codec init\n");
++	codec->name = "UDA1380";
++	codec->owner = THIS_MODULE;
++	codec->read = uda1380_read_reg_cache;
++	codec->write = uda1380_write;
++	codec->dapm_event = uda1380_dapm_event;
++	codec->dai = uda1380_dai;
++	codec->num_dai = ARRAY_SIZE(uda1380_dai);
++	codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
++	codec->reg_cache =
++			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);
++	/* set clock input */
++	switch (dac_clk) {
++	case UDA1380_DAC_CLK_SYSCLK:
++		uda1380_write(codec, UDA1380_CLK, 0);
++		break;
++	case UDA1380_DAC_CLK_WSPLL:
++		uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
++		break;
++	}
++
++	/* uda1380 init */
++	uda1380_add_controls(codec);
++	uda1380_add_widgets(codec);
++	ret = snd_soc_register_card(socdev);
++	if(ret < 0) {
++		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, setup->dac_clk);
++	if(ret < 0) {
++		printk(KERN_ERR "failed to initialise UDA1380\n");
++		goto err;
++	}
++	return ret;
++
++err:
++	kfree(codec);
++	kfree(i2c);
++	return ret;
++}
++
++static int uda1380_i2c_detach(struct i2c_client *client)
++{
++	struct snd_soc_codec* codec = i2c_get_clientdata(client);
++	i2c_detach_client(client);
++	kfree(codec->reg_cache);
++	kfree(client);
++	return 0;
++}
++
++static int uda1380_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, uda1380_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver uda1380_i2c_driver = {
++	.driver = {
++		.name =  "UDA1380 I2C Codec",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_UDA1380,
++	.attach_adapter = uda1380_i2c_attach,
++	.detach_client =  uda1380_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "UDA1380",
++	.driver = &uda1380_i2c_driver,
++};
++#endif
++
++static int uda1380_probe(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct uda1380_setup_data *setup;
++	struct snd_soc_codec* codec;
++	int ret = 0;
++
++	printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION);
++
++	setup = socdev->codec_data;
++	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++	if (codec == NULL)
++		return -ENOMEM;
++
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++
++	uda1380_socdev = socdev;
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++printk("uda1380 i2c reg\n");
++	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-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,84 @@
++/*
++ * Audio support for Philips UDA1380
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ * Copyright (c) 2005 Giorgio Padrin <giorgio at mandarinlogiq.org>
++ */
++
++#define UDA1380_CLK		0x00
++#define UDA1380_IFACE	0x01
++#define UDA1380_PM		0x02
++#define UDA1380_AMIX	0x03
++#define UDA1380_HP		0x04
++#define UDA1380_MVOL	0x10
++#define UDA1380_MIXVOL	0x11
++#define UDA1380_MODE	0x12
++#define UDA1380_DEEMP	0x13
++#define UDA1380_MIXER	0x14
++#define UDA1380_INTSTAT	0x18
++#define UDA1380_DEC		0x20
++#define UDA1380_PGA		0x21
++#define UDA1380_ADC		0x22
++#define UDA1380_AGC		0x23
++#define UDA1380_DECSTAT	0x28
++#define UDA1380_RESET	0x7f
++
++#define UDA1380_CACHEREGNUM 0x24
++
++/* Register flags */
++#define R00_EN_ADC	0x0800
++#define R00_EN_DEC	0x0400
++#define R00_EN_DAC	0x0200
++#define R00_EN_INT	0x0100
++#define R00_DAC_CLK	0x0010
++#define R01_SFORI_I2S   0x0000
++#define R01_SFORI_LSB16 0x0100
++#define R01_SFORI_LSB18 0x0200
++#define R01_SFORI_LSB20 0x0300
++#define R01_SFORI_MSB   0x0500
++#define R01_SFORI_MASK  0x0700
++#define R01_SFORO_I2S   0x0000
++#define R01_SFORO_LSB16 0x0001
++#define R01_SFORO_LSB18 0x0002
++#define R01_SFORO_LSB20 0x0003
++#define R01_SFORO_LSB24 0x0004
++#define R01_SFORO_MSB   0x0005
++#define R01_SFORO_MASK  0x0007
++#define R01_SEL_SOURCE  0x0040
++#define R01_SIM		0x0010
++#define R02_PON_PLL	0x8000
++#define R02_PON_HP	0x2000
++#define R02_PON_DAC	0x0400
++#define R02_PON_BIAS	0x0100
++#define R02_EN_AVC	0x0080
++#define R02_PON_AVC	0x0040
++#define R02_PON_LNA	0x0010
++#define R02_PON_PGAL	0x0008
++#define R02_PON_ADCL	0x0004
++#define R02_PON_PGAR	0x0002
++#define R02_PON_ADCR	0x0001
++#define R13_MTM		0x4000
++#define R14_SILENCE	0x0080
++#define R14_SDET_ON	0x0040
++#define R21_MT_ADC	0x8000
++#define R22_SEL_LNA	0x0008
++#define R22_SEL_MIC	0x0004
++#define R22_SKIP_DCFIL	0x0002
++#define R23_AGC_EN	0x0001
++
++struct uda1380_setup_data {
++	unsigned short i2c_address;
++	int            dac_clk;
++#define UDA1380_DAC_CLK_SYSCLK 0
++#define UDA1380_DAC_CLK_WSPLL  1
++};
++
++#define UDA1380_DAI_DUPLEX		0 /* playback and capture on single DAI */
++#define UDA1380_DAI_PLAYBACK	1 /* playback DAI */
++#define UDA1380_DAI_CAPTURE		2 /* capture DAI */
++
++extern struct snd_soc_codec_dai uda1380_dai[3];
++extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,757 @@
++/*
++ * wm8731.c  --  WM8731 ALSA SoC Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard at openedhand.com>
++ *
++ * Based on wm8753.c by Liam Girdwood
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8731.h"
++
++#define AUDIO_NAME "wm8731"
++#define WM8731_VERSION "0.13"
++
++/*
++ * 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;
++
++/* codec private data */
++struct wm8731_priv {
++	unsigned int sysclk;
++};
++
++/*
++ * 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
++};
++
++/*
++ * 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},
++	{12000000, 32000, 375, 0x6, 0x0, 0x1},
++
++	/* 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;
++}
++
++static int wm8731_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8731_priv *wm8731 = codec->private_data;
++	u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
++	int i = get_coeff(wm8731->sysclk, snd_soc_get_rate(params_rate(params)));
++	u16 srate = (coeff_div[i].sr << 2) |
++		(coeff_div[i].bosr << 1) | coeff_div[i].usb;
++
++	wm8731_write(codec, WM8731_SRATE, srate);
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0008;
++		break;
++	}
++
++	wm8731_write(codec, WM8731_IFACE, iface);
++	return 0;
++}
++
++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;
++
++	/* 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_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	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_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct wm8731_priv *wm8731 = codec->private_data;
++
++	switch (freq) {
++	case 11289600:
++	case 12000000:
++	case 12288000:
++	case 16934400:
++	case 18432000:
++		wm8731->sysclk = freq;
++		return 0;
++	}
++	return -EINVAL;
++}
++
++
++static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = 0;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		iface |= 0x0040;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	/* set iface */
++	wm8731_write(codec, WM8731_IFACE, iface);
++	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;
++}
++
++#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_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_codec_dai wm8731_dai = {
++	.name = "WM8731",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8731_RATES,
++		.formats = WM8731_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8731_RATES,
++		.formats = WM8731_FORMATS,},
++	.ops = {
++		.prepare = wm8731_pcm_prepare,
++		.hw_params = wm8731_hw_params,
++		.shutdown = wm8731_shutdown,
++	},
++	.dai_ops = {
++		.digital_mute = wm8731_mute,
++		.set_sysclk = wm8731_set_dai_sysclk,
++		.set_fmt = wm8731_set_dai_fmt,
++	}
++};
++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;
++	struct wm8731_priv *wm8731;
++	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;
++
++	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
++	if (wm8731 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = wm8731;
++	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->private_data);
++	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-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,44 @@
++/*
++ * wm8731.h  --  WM8731 Soc Audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard at openedhand.com>
++ *
++ * Based on wm8753.h
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ */
++
++#ifndef _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
++
++#define WM8731_SYSCLK	0
++#define WM8731_DAI		0
++
++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-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,1039 @@
++/*
++ * wm8750.c -- WM8750 ALSA SoC audio driver
++ *
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard at 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 "0.12"
++
++/*
++ * 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;
++
++/* codec private data */
++struct wm8750_priv {
++	unsigned int sysclk;
++};
++
++/*
++ * 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 */
++};
++
++/*
++ * 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;
++}
++
++static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct wm8750_priv *wm8750 = codec->private_data;
++
++	switch (freq) {
++	case 11289600:
++	case 12000000:
++	case 12288000:
++	case 16934400:
++	case 18432000:
++		wm8750->sysclk = freq;
++		return 0;
++	}
++	return -EINVAL;
++}
++
++static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = 0;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		iface = 0x0040;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	wm8750_write(codec, WM8750_IFACE, iface);
++	return 0;
++}
++
++static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8750_priv *wm8750 = codec->private_data;
++	u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
++	u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
++	int coeff = get_coeff(wm8750->sysclk, params_rate(params));
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0008;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		iface |= 0x000c;
++		break;
++	}
++
++	/* set iface & srate */
++	wm8750_write(codec, WM8750_IFACE, iface);
++	if (coeff >= 0)
++		wm8750_write(codec, WM8750_SRATE, srate |
++			(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
++
++	return 0;
++}
++
++static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	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;
++}
++
++#define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_codec_dai wm8750_dai = {
++	.name = "WM8750",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8750_RATES,
++		.formats = WM8750_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8750_RATES,
++		.formats = WM8750_FORMATS,},
++	.ops = {
++		.hw_params = wm8750_pcm_hw_params,
++	},
++	.dai_ops = {
++		.digital_mute = wm8750_mute,
++		.set_fmt = wm8750_set_dai_fmt,
++		.set_sysclk = wm8750_set_dai_sysclk,
++	},
++};
++EXPORT_SYMBOL_GPL(wm8750_dai);
++
++static void wm8750_work(struct work_struct *work)
++{
++	struct snd_soc_codec *codec =
++		container_of(work, struct snd_soc_codec, delayed_work.work);
++	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, &codec->delayed_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, &codec->delayed_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;
++	struct wm8750_priv *wm8750;
++	int ret = 0;
++
++	info("WM8750 Audio Codec %s", WM8750_VERSION);
++	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++	if (codec == NULL)
++		return -ENOMEM;
++
++	wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
++	if (wm8750 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = wm8750;
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++	wm8750_socdev = socdev;
++	INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
++	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->private_data);
++	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-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,67 @@
++/*
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Richard Purdie <richard at openedhand.com>
++ *
++ * Based on WM8753.h
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ */
++
++#ifndef _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
++
++#define WM8750_SYSCLK	0
++
++struct wm8750_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8750_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8750;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,1714 @@
++/*
++ * wm8753.c  --  WM8753 ALSA Soc Audio driver
++ *
++ * Copyright 2003 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ * Notes:
++ *  The WM8753 is a low power, high quality stereo codec with integrated PCM
++ *  codec designed for portable digital telephony applications.
++ *
++ * Dual DAI:-
++ *
++ * This driver support 2 DAI PCM's. This makes the default PCM available for
++ * HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for
++ * voice.
++ *
++ * Please note that the voice PCM can be connected directly to a Bluetooth
++ * codec or GSM modem and thus cannot be read or written to, although it is
++ * available to be configured with snd_hw_params(), etc and kcontrols in the
++ * normal alsa manner.
++ *
++ * Fast DAI switching:-
++ *
++ * The driver can now fast switch between the DAI configurations via a
++ * an alsa kcontrol. This allows the PCM to remain open.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++#include <asm/div64.h>
++
++#include "wm8753.h"
++
++#define AUDIO_NAME "wm8753"
++#define WM8753_VERSION "0.16"
++
++/*
++ * Debug
++ */
++
++#define WM8753_DEBUG 0
++
++#ifdef WM8753_DEBUG
++#define dbg(format, arg...) \
++	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++static int caps_charge = 2000;
++module_param(caps_charge, int, 0);
++MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
++
++static struct workqueue_struct *wm8753_workq = NULL;
++static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
++	unsigned int mode);
++
++/* codec private data */
++struct wm8753_priv {
++	unsigned int sysclk;
++	unsigned int pcmclk;
++};
++
++/*
++ * wm8753 register cache
++ * We can't read the WM8753 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8753_reg[] = {
++	0x0008, 0x0000, 0x000a, 0x000a,
++	0x0033, 0x0000, 0x0007, 0x00ff,
++	0x00ff, 0x000f, 0x000f, 0x007b,
++	0x0000, 0x0032, 0x0000, 0x00c3,
++	0x00c3, 0x00c0, 0x0000, 0x0000,
++	0x0000, 0x0000, 0x0000, 0x0000,
++	0x0000, 0x0000, 0x0000, 0x0000,
++	0x0000, 0x0000, 0x0000, 0x0055,
++	0x0005, 0x0050, 0x0055, 0x0050,
++	0x0055, 0x0050, 0x0055, 0x0079,
++	0x0079, 0x0079, 0x0079, 0x0079,
++	0x0000, 0x0000, 0x0000, 0x0000,
++	0x0097, 0x0097, 0x0000, 0x0004,
++	0x0000, 0x0083, 0x0024, 0x01ba,
++	0x0000, 0x0083, 0x0024, 0x01ba,
++	0x0000, 0x0000
++};
++
++/*
++ * read wm8753 register cache
++ */
++static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
++		return -1;
++	return cache[reg - 1];
++}
++
++/*
++ * write wm8753 register cache
++ */
++static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg < 1 || reg > 0x3f)
++		return;
++	cache[reg - 1] = value;
++}
++
++/*
++ * write to the WM8753 register space
++ */
++static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D9 WM8753 register offset
++	 *   D8...D0 register data
++	 */
++	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++	data[1] = value & 0x00ff;
++
++	wm8753_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -EIO;
++}
++
++#define wm8753_reset(c) wm8753_write(c, WM8753_RESET, 0)
++
++/*
++ * WM8753 Controls
++ */
++static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"};
++static const char *wm8753_base_filter[] =
++	{"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz",
++	"100Hz @ 8kHz", "200Hz @ 8kHz"};
++static const char *wm8753_treble[] = {"8kHz", "4kHz"};
++static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"};
++static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"};
++static const char *wm8753_3d_func[] = {"Capture", "Playback"};
++static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"};
++static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"};
++static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"};
++static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"};
++static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"};
++static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2",
++	"Line 1", "Line 2"};
++static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"};
++static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"};
++static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"};
++static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"};
++static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2",
++	"Right PGA"};
++static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right",
++	"Left + Right"};
++static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"};
++static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"};
++static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"};
++static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"};
++static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left",
++	"Analogue Mix Right", "Digital Mono Mix"};
++static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k",
++	"82Hz @ 8kHz", "170Hz @ 8kHz"};
++static const char *wm8753_adc_filter[] = {"HiFi", "Voice"};
++static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"};
++static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"};
++
++static const 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, 255, 0),
++
++SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0),
++SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0),
++
++SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0),
++
++SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1),
++SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1),
++SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1),
++
++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0),
++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0),
++
++SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
++SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
++SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1),
++SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
++
++SOC_ENUM("Bass Boost", wm8753_enum[0]),
++SOC_ENUM("Bass Filter", wm8753_enum[1]),
++SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 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, 1),
++
++SOC_ENUM("Capture Filter Select", wm8753_enum[23]),
++SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]),
++SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1),
++
++SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0),
++SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0),
++SOC_ENUM("ALC Capture Function", wm8753_enum[3]),
++SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0),
++SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1),
++SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1),
++SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0),
++SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0),
++SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]),
++SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0),
++
++SOC_ENUM("3D Function", wm8753_enum[5]),
++SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]),
++SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]),
++SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0),
++SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0),
++
++SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0),
++SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0),
++
++SOC_ENUM("De-emphasis", wm8753_enum[8]),
++SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
++SOC_ENUM("Playback Phase", wm8753_enum[10]),
++
++SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0),
++SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
++
++SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
++};
++
++/* add non dapm controls */
++static int wm8753_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++	return 0;
++}
++
++/*
++ * _DAPM_ Controls
++ */
++
++/* Left Mixer */
++static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0),
++SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0),
++};
++
++/* Right mixer */
++static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0),
++};
++
++/* Mono mixer */
++static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0),
++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0),
++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0),
++};
++
++/* Mono 2 Mux */
++static const struct snd_kcontrol_new wm8753_mono2_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[17]);
++
++/* Out 3 Mux */
++static const struct snd_kcontrol_new wm8753_out3_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[18]);
++
++/* Out 4 Mux */
++static const struct snd_kcontrol_new wm8753_out4_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[19]);
++
++/* ADC Mono Mix */
++static const struct snd_kcontrol_new wm8753_adc_mono_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[22]);
++
++/* Record mixer */
++static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = {
++SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0),
++SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0),
++SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0),
++};
++
++/* Left ADC mux */
++static const struct snd_kcontrol_new wm8753_adc_left_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[21]);
++
++/* Right ADC mux */
++static const struct snd_kcontrol_new wm8753_adc_right_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[20]);
++
++/* MIC mux */
++static const struct snd_kcontrol_new wm8753_mic_mux_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[16]);
++
++/* ALC mixer */
++static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0),
++SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0),
++SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0),
++SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0),
++};
++
++/* Left Line mux */
++static const struct snd_kcontrol_new wm8753_line_left_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[14]);
++
++/* Right Line mux */
++static const struct snd_kcontrol_new wm8753_line_right_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[13]);
++
++/* Mono Line mux */
++static const struct snd_kcontrol_new wm8753_line_mono_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[12]);
++
++/* Line mux and mixer */
++static const struct snd_kcontrol_new wm8753_line_mux_mix_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[11]);
++
++/* Rx mux and mixer */
++static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[15]);
++
++/* Mic Selector Mux */
++static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls =
++SOC_DAPM_ENUM("Route", wm8753_enum[25]);
++
++static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0),
++SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0,
++	&wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)),
++SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0),
++SND_SOC_DAPM_OUTPUT("LOUT1"),
++SND_SOC_DAPM_OUTPUT("LOUT2"),
++SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0,
++	&wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)),
++SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0),
++SND_SOC_DAPM_OUTPUT("ROUT1"),
++SND_SOC_DAPM_OUTPUT("ROUT2"),
++SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0,
++	&wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)),
++SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0),
++SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0),
++SND_SOC_DAPM_OUTPUT("MONO1"),
++SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls),
++SND_SOC_DAPM_OUTPUT("MONO2"),
++SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0),
++SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls),
++SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0),
++SND_SOC_DAPM_OUTPUT("OUT3"),
++SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls),
++SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0),
++SND_SOC_DAPM_OUTPUT("OUT4"),
++SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0,
++	&wm8753_record_mixer_controls[0],
++	ARRAY_SIZE(wm8753_record_mixer_controls)),
++SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8753_PWR2, 3, 0),
++SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8753_PWR2, 2, 0),
++SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0,
++	&wm8753_adc_mono_controls),
++SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0,
++	&wm8753_adc_mono_controls),
++SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0,
++	&wm8753_adc_left_controls),
++SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0,
++	&wm8753_adc_right_controls),
++SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0,
++	&wm8753_mic_mux_controls),
++SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0),
++SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0,
++	&wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)),
++SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0,
++	&wm8753_line_left_controls),
++SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0,
++	&wm8753_line_right_controls),
++SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0,
++	&wm8753_line_mono_controls),
++SND_SOC_DAPM_MUX("Line Mixer", 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_OUTPUT("ACOP"),
++SND_SOC_DAPM_INPUT("MIC1N"),
++SND_SOC_DAPM_INPUT("MIC1"),
++SND_SOC_DAPM_INPUT("MIC2N"),
++SND_SOC_DAPM_INPUT("MIC2"),
++SND_SOC_DAPM_VMID("VREF"),
++};
++
++static const char *audio_map[][3] = {
++	/* left mixer */
++	{"Left Mixer", "Left Playback Switch", "Left DAC"},
++	{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
++	{"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
++	{"Left Mixer", "Bypass Playback Switch", "Line Left Mux"},
++
++	/* right mixer */
++	{"Right Mixer", "Right Playback Switch", "Right DAC"},
++	{"Right Mixer", "Voice Playback Switch", "Voice DAC"},
++	{"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
++	{"Right Mixer", "Bypass Playback Switch", "Line Right Mux"},
++
++	/* mono mixer */
++	{"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
++	{"Mono Mixer", "Left Playback Switch", "Left DAC"},
++	{"Mono Mixer", "Right Playback Switch", "Right DAC"},
++	{"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
++	{"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"},
++
++	/* left out */
++	{"Left Out 1", NULL, "Left Mixer"},
++	{"Left Out 2", NULL, "Left Mixer"},
++	{"LOUT1", NULL, "Left Out 1"},
++	{"LOUT2", NULL, "Left Out 2"},
++
++	/* right out */
++	{"Right Out 1", NULL, "Right Mixer"},
++	{"Right Out 2", NULL, "Right Mixer"},
++	{"ROUT1", NULL, "Right Out 1"},
++	{"ROUT2", NULL, "Right Out 2"},
++
++	/* mono 1 out */
++	{"Mono Out 1", NULL, "Mono Mixer"},
++	{"MONO1", NULL, "Mono Out 1"},
++
++	/* mono 2 out */
++	{"Mono 2 Mux", "Left + Right", "Out3 Left + Right"},
++	{"Mono 2 Mux", "Inverted Mono 1", "MONO1"},
++	{"Mono 2 Mux", "Left", "Left Mixer"},
++	{"Mono 2 Mux", "Right", "Right Mixer"},
++	{"Mono Out 2", NULL, "Mono 2 Mux"},
++	{"MONO2", NULL, "Mono Out 2"},
++
++	/* out 3 */
++	{"Out3 Left + Right", NULL, "Left Mixer"},
++	{"Out3 Left + Right", NULL, "Right Mixer"},
++	{"Out3 Mux", "VREF", "VREF"},
++	{"Out3 Mux", "Left + Right", "Out3 Left + Right"},
++	{"Out3 Mux", "ROUT2", "ROUT2"},
++	{"Out 3", NULL, "Out3 Mux"},
++	{"OUT3", NULL, "Out 3"},
++
++	/* out 4 */
++	{"Out4 Mux", "VREF", "VREF"},
++	{"Out4 Mux", "Capture ST", "Capture ST Mixer"},
++	{"Out4 Mux", "LOUT2", "LOUT2"},
++	{"Out 4", NULL, "Out4 Mux"},
++	{"OUT4", NULL, "Out 4"},
++
++	/* record mixer  */
++	{"Playback Mixer", "Left Capture Switch", "Left Mixer"},
++	{"Playback Mixer", "Voice Capture Switch", "Mono Mixer"},
++	{"Playback Mixer", "Right Capture Switch", "Right Mixer"},
++
++	/* Mic/SideTone Mux */
++	{"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"},
++	{"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"},
++	{"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"},
++	{"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"},
++
++	/* Capture Left Mux */
++	{"Capture Left Mux", "PGA", "Left Capture Volume"},
++	{"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"},
++	{"Capture Left Mux", "Line", "LINE1"},
++
++	/* Capture Right Mux */
++	{"Capture Right Mux", "PGA", "Right Capture Volume"},
++	{"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
++	{"Capture Right Mux", "Sidetone", "Capture ST Mixer"},
++
++	/* Mono Capture mixer-mux */
++	{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
++	{"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
++	{"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
++	{"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
++	{"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"},
++	{"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"},
++	{"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"},
++	{"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"},
++	{"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"},
++
++	/* ADC */
++	{"Left ADC", NULL, "Capture Left Mixer"},
++	{"Right ADC", NULL, "Capture Right Mixer"},
++
++	/* Left Capture Volume */
++	{"Left Capture Volume", NULL, "ACIN"},
++
++	/* Right Capture Volume */
++	{"Right Capture Volume", NULL, "Mic 2 Volume"},
++
++	/* ALC Mixer */
++	{"ALC Mixer", "Line Capture Switch", "Line Mixer"},
++	{"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"},
++	{"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"},
++	{"ALC Mixer", "Rx Capture Switch", "Rx Mixer"},
++
++	/* Line Left Mux */
++	{"Line Left Mux", "Line 1", "LINE1"},
++	{"Line Left Mux", "Rx Mix", "Rx Mixer"},
++
++	/* Line Right Mux */
++	{"Line Right Mux", "Line 2", "LINE2"},
++	{"Line Right Mux", "Rx Mix", "Rx Mixer"},
++
++	/* Line Mono Mux */
++	{"Line Mono Mux", "Line Mix", "Line Mixer"},
++	{"Line Mono Mux", "Rx Mix", "Rx Mixer"},
++
++	/* Line Mixer/Mux */
++	{"Line Mixer", "Line 1 + 2", "LINE1"},
++	{"Line Mixer", "Line 1 - 2", "LINE1"},
++	{"Line Mixer", "Line 1 + 2", "LINE2"},
++	{"Line Mixer", "Line 1 - 2", "LINE2"},
++	{"Line Mixer", "Line 1", "LINE1"},
++	{"Line Mixer", "Line 2", "LINE2"},
++
++	/* Rx Mixer/Mux */
++	{"Rx Mixer", "RXP - RXN", "RXP"},
++	{"Rx Mixer", "RXP + RXN", "RXP"},
++	{"Rx Mixer", "RXP - RXN", "RXN"},
++	{"Rx Mixer", "RXP + RXN", "RXN"},
++	{"Rx Mixer", "RXP", "RXP"},
++	{"Rx Mixer", "RXN", "RXN"},
++
++	/* Mic 1 Volume */
++	{"Mic 1 Volume", NULL, "MIC1N"},
++	{"Mic 1 Volume", NULL, "Mic Selection Mux"},
++
++	/* Mic 2 Volume */
++	{"Mic 2 Volume", NULL, "MIC2N"},
++	{"Mic 2 Volume", NULL, "MIC2"},
++
++	/* Mic Selector Mux */
++	{"Mic Selection Mux", "Mic 1", "MIC1"},
++	{"Mic Selection Mux", "Mic 2", "MIC2N"},
++	{"Mic Selection Mux", "Mic 3", "MIC2"},
++
++	/* ACOP */
++	{"ACOP", NULL, "ALC Mixer"},
++
++	/* terminator */
++	{NULL, NULL, NULL},
++};
++
++static int wm8753_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
++	}
++
++	/* set up the WM8753 audio map */
++	for(i = 0; audio_map[i][0] != NULL; i++) {
++		snd_soc_dapm_connect_input(codec, audio_map[i][0],
++			audio_map[i][1], audio_map[i][2]);
++	}
++
++	snd_soc_dapm_new_widgets(codec);
++	return 0;
++}
++
++/* PLL divisors */
++struct _pll_div {
++	u32 div2:1;
++	u32 n:4;
++	u32 k:24;
++};
++
++static struct _pll_div pll_div;
++
++/* The size in bits of the pll divide multiplied by 10
++ * to allow rounding later */
++#define FIXED_PLL_SIZE ((1 << 22) * 10)
++
++static void pll_factors(unsigned int target, unsigned int source)
++{
++	unsigned long long Kpart;
++	unsigned int K, Ndiv, Nmod;
++
++	Ndiv = target / source;
++	if (Ndiv < 6) {
++		source >>= 1;
++		pll_div.div2 = 1;
++		Ndiv = target / source;
++	} else
++		pll_div.div2 = 0;
++
++	if ((Ndiv < 6) || (Ndiv > 12))
++		printk(KERN_WARNING
++			"WM8753 N value outwith recommended range! N = %d\n",Ndiv);
++
++	pll_div.n = Ndiv;
++	Nmod = target % source;
++	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
++
++	do_div(Kpart, source);
++
++	K = Kpart & 0xFFFFFFFF;
++
++	/* Check if we need to round */
++	if ((K % 10) >= 5)
++		K += 5;
++
++	/* Move down to proper range now rounding is done */
++	K /= 10;
++
++	pll_div.k = K;
++}
++
++static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	u16 reg, enable;
++	int offset;
++	struct snd_soc_codec *codec = codec_dai->codec;
++
++	if (pll_id < WM8753_PLL1 || pll_id > WM8753_PLL2)
++		return -ENODEV;
++
++	if (pll_id == WM8753_PLL1) {
++		offset = 0;
++		enable = 0x10;
++		reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xffef;
++	} else {
++		offset = 4;
++		enable = 0x8;
++		reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfff7;
++	}
++
++	if (!freq_in || !freq_out) {
++		/* disable PLL  */
++		wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0026);
++		wm8753_write(codec, WM8753_CLOCK, reg);
++		return 0;
++	} else {
++
++        u16 value = 0;
++
++		pll_factors(freq_out * 8, freq_in);
++
++        /* set up N and K PLL divisor ratios */
++        /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
++        value = (pll_div.n << 5) + ((pll_div.k & 0x3c0000) >> 18);
++        wm8753_write(codec, WM8753_PLL1CTL2 + offset, value);
++
++        /* bits 8:0 = PLL_K[17:9] */
++        value = (pll_div.k & 0x03fe00) >> 9;
++        wm8753_write(codec, WM8753_PLL1CTL3 + offset, value);
++
++        /* bits 8:0 = PLL_K[8:0] */
++        value = pll_div.k & 0x0001ff;
++        wm8753_write(codec, WM8753_PLL1CTL4 + offset, value);
++
++        /* set PLL as input and enable */
++        wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0027 |
++        	(pll_div.div2 << 3));
++		wm8753_write(codec, WM8753_CLOCK, reg | enable);
++    }
++	return 0;
++}
++
++struct _coeff_div {
++	u32 mclk;
++	u32 rate;
++	u8 sr:5;
++	u8 usb:1;
++};
++
++/* codec hifi mclk (after PLL) clock divider coefficients */
++static const struct _coeff_div coeff_div[] = {
++	/* 8k */
++	{12288000, 8000, 0x6, 0x0},
++	{11289600, 8000, 0x16, 0x0},
++	{18432000, 8000, 0x7, 0x0},
++	{16934400, 8000, 0x17, 0x0},
++	{12000000, 8000, 0x6, 0x1},
++
++	/* 11.025k */
++	{11289600, 11025, 0x18, 0x0},
++	{16934400, 11025, 0x19, 0x0},
++	{12000000, 11025, 0x19, 0x1},
++
++	/* 16k */
++	{12288000, 16000, 0xa, 0x0},
++	{18432000, 16000, 0xb, 0x0},
++	{12000000, 16000, 0xa, 0x1},
++
++	/* 22.05k */
++	{11289600, 22050, 0x1a, 0x0},
++	{16934400, 22050, 0x1b, 0x0},
++	{12000000, 22050, 0x1b, 0x1},
++
++	/* 32k */
++	{12288000, 32000, 0xc, 0x0},
++	{18432000, 32000, 0xd, 0x0},
++	{12000000, 32000, 0xa, 0x1},
++
++	/* 44.1k */
++	{11289600, 44100, 0x10, 0x0},
++	{16934400, 44100, 0x11, 0x0},
++	{12000000, 44100, 0x11, 0x1},
++
++	/* 48k */
++	{12288000, 48000, 0x0, 0x0},
++	{18432000, 48000, 0x1, 0x0},
++	{12000000, 48000, 0x0, 0x1},
++
++	/* 88.2k */
++	{11289600, 88200, 0x1e, 0x0},
++	{16934400, 88200, 0x1f, 0x0},
++	{12000000, 88200, 0x1f, 0x1},
++
++	/* 96k */
++	{12288000, 96000, 0xe, 0x0},
++	{18432000, 96000, 0xf, 0x0},
++	{12000000, 96000, 0xe, 0x1},
++};
++
++static int get_coeff(int mclk, int rate)
++{
++	int i;
++
++	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
++		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
++			return i;
++	}
++	return -EINVAL;
++}
++
++/*
++ * Clock after PLL and dividers
++ */
++static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct wm8753_priv *wm8753 = codec->private_data;
++
++	switch (freq) {
++	case 11289600:
++	case 12000000:
++	case 12288000:
++	case 16934400:
++	case 18432000:
++		if (clk_id == WM8753_MCLK) {
++			wm8753->sysclk = freq;
++			return 0;
++		} else if (clk_id == WM8753_PCMCLK) {
++			wm8753->pcmclk = freq;
++			return 0;
++		}
++	}
++	return -EINVAL;
++}
++
++/*
++ * Set's ADC and Voice DAC format.
++ */
++static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01ec;
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		voice |= 0x0002;
++		break;
++	case SND_SOC_DAIFMT_RIGHT_J:
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		voice |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_DSP_A:
++		voice |= 0x0003;
++		break;
++	case SND_SOC_DAIFMT_DSP_B:
++		voice |= 0x0013;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	wm8753_write(codec, WM8753_PCM, voice);
++	return 0;
++}
++
++/*
++ * Set PCM DAI bit size and sample rate.
++ */
++static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8753_priv *wm8753 = codec->private_data;
++	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
++	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		voice |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		voice |= 0x0008;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		voice |= 0x000c;
++		break;
++	}
++
++	/* sample rate */
++	if (params_rate(params) * 384 == wm8753->pcmclk)
++		srate |= 0x80;
++	wm8753_write(codec, WM8753_SRATE1, srate);
++
++	wm8753_write(codec, WM8753_PCM, voice);
++	return 0;
++}
++
++/*
++ * Set's PCM dai fmt and BCLK.
++ */
++static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 voice, ioctl;
++
++	voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x010f;
++	ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x010d;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	case SND_SOC_DAIFMT_CBM_CFM:
++		ioctl |= 0x2;
++	case SND_SOC_DAIFMT_CBM_CFS:
++		voice |= 0x0040;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
++	case SND_SOC_DAIFMT_NB_NF:
++		break;
++	case SND_SOC_DAIFMT_IB_IF:
++		voice |= 0x0090;
++		break;
++	case SND_SOC_DAIFMT_IB_NF:
++		voice |= 0x0080;
++		break;
++	case SND_SOC_DAIFMT_NB_IF:
++		voice |= 0x0010;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	wm8753_write(codec, WM8753_PCM, voice);
++	wm8753_write(codec, WM8753_IOCTL, ioctl);
++	return 0;
++}
++
++static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	switch (div_id) {
++	case WM8753_PCMDIV:
++		reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0x003f;
++		wm8753_write(codec, WM8753_CLOCK, reg | div);
++		break;
++	case WM8753_BCLKDIV:
++		reg = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x01c7;
++		wm8753_write(codec, WM8753_SRATE2, reg | div);
++		break;
++	case WM8753_VXCLKDIV:
++		reg = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x003f;
++		wm8753_write(codec, WM8753_SRATE2, reg | div);
++		break;
++	default:
++		return -EINVAL;
++	}
++	return 0;
++}
++
++/*
++ * Set's HiFi DAC format.
++ */
++static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01e0;
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		hifi |= 0x0002;
++		break;
++	case SND_SOC_DAIFMT_RIGHT_J:
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		hifi |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_DSP_A:
++		hifi |= 0x0003;
++		break;
++	case SND_SOC_DAIFMT_DSP_B:
++		hifi |= 0x0013;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	wm8753_write(codec, WM8753_HIFI, hifi);
++	return 0;
++}
++
++/*
++ * Set's I2S DAI format.
++ */
++static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 ioctl, hifi;
++
++	hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x010f;
++	ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x00fe;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	case SND_SOC_DAIFMT_CBM_CFM:
++		ioctl |= 0x1;
++	case SND_SOC_DAIFMT_CBM_CFS:
++		hifi |= 0x0040;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
++	case SND_SOC_DAIFMT_NB_NF:
++		break;
++	case SND_SOC_DAIFMT_IB_IF:
++		hifi |= 0x0090;
++		break;
++	case SND_SOC_DAIFMT_IB_NF:
++		hifi |= 0x0080;
++		break;
++	case SND_SOC_DAIFMT_NB_IF:
++		hifi |= 0x0010;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	wm8753_write(codec, WM8753_HIFI, hifi);
++	wm8753_write(codec, WM8753_IOCTL, ioctl);
++	return 0;
++}
++
++/*
++ * Set PCM DAI bit size and sample rate.
++ */
++static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8753_priv *wm8753 = codec->private_data;
++	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
++	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
++	int coeff;
++
++	/* is digital filter coefficient valid ? */
++	coeff = get_coeff(wm8753->sysclk, params_rate(params));
++	if (coeff < 0) {
++		printk(KERN_ERR "wm8753 invalid MCLK or rate\n");
++		return coeff;
++	}
++	wm8753_write(codec, WM8753_SRATE1, srate | (coeff_div[coeff].sr << 1) |
++		coeff_div[coeff].usb);
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		hifi |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		hifi |= 0x0008;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		hifi |= 0x000c;
++		break;
++	}
++
++	wm8753_write(codec, WM8753_HIFI, hifi);
++	return 0;
++}
++
++static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 clock;
++
++	/* set clk source as pcmclk */
++	clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++	wm8753_write(codec, WM8753_CLOCK, clock);
++
++	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
++		return -EINVAL;
++	return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
++}
++
++static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
++		return -EINVAL;
++	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
++}
++
++static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 clock;
++
++	/* set clk source as pcmclk */
++	clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++	wm8753_write(codec, WM8753_CLOCK, clock);
++
++	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
++		return -EINVAL;
++	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
++}
++
++static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 clock;
++
++	/* set clk source as mclk */
++	clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
++	wm8753_write(codec, WM8753_CLOCK, clock | 0x4);
++
++	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
++		return -EINVAL;
++	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
++		return -EINVAL;
++	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
++}
++
++static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
++
++	/* the digital mute covers the HiFi and Voice DAC's on the WM8753.
++	 * make sure we check if they are not both active when we mute */
++	if (mute && dai->id == 1) {
++		if (!wm8753_dai[WM8753_DAI_VOICE].playback.active ||
++			!wm8753_dai[WM8753_DAI_HIFI].playback.active)
++			wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
++	} else {
++		if (mute)
++			wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
++		else
++			wm8753_write(codec, WM8753_DAC, mute_reg);
++	}
++
++	return 0;
++}
++
++static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
++{
++	u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
++
++	switch (event) {
++	case SNDRV_CTL_POWER_D0: /* full On */
++		/* set vmid to 50k and unmute dac */
++		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
++		break;
++	case SNDRV_CTL_POWER_D1: /* partial On */
++	case SNDRV_CTL_POWER_D2: /* partial On */
++		/* set vmid to 5k for quick power up */
++		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
++		break;
++	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++		/* mute dac and set vmid to 500k, enable VREF */
++		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
++		break;
++	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++		wm8753_write(codec, WM8753_PWR1, 0x0001);
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define WM8753_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8753_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++/*
++ * The WM8753 supports upto 4 different and mutually exclusive DAI
++ * configurations. This gives 2 PCM's available for use, hifi and voice.
++ * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
++ * is connected between the wm8753 and a BT codec or GSM modem.
++ *
++ * 1. Voice over PCM DAI - HIFI DAC over HIFI DAI
++ * 2. Voice over HIFI DAI - HIFI disabled
++ * 3. Voice disabled - HIFI over HIFI
++ * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
++ */
++static const struct snd_soc_codec_dai wm8753_all_dai[] = {
++/* DAI HiFi mode 1 */
++{	.name = "WM8753 HiFi",
++	.id = 1,
++	.playback = {
++		.stream_name = "HiFi Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.capture = { /* dummy for fast DAI switching */
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.ops = {
++		.hw_params = wm8753_i2s_hw_params,},
++	.dai_ops = {
++		.digital_mute = wm8753_mute,
++		.set_fmt = wm8753_mode1h_set_dai_fmt,
++		.set_clkdiv = wm8753_set_dai_clkdiv,
++		.set_pll = wm8753_set_dai_pll,
++		.set_sysclk = wm8753_set_dai_sysclk,
++	},
++},
++/* DAI Voice mode 1 */
++{	.name = "WM8753 Voice",
++	.id = 1,
++	.playback = {
++		.stream_name = "Voice Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.ops = {
++		.hw_params = wm8753_pcm_hw_params,},
++	.dai_ops = {
++		.digital_mute = wm8753_mute,
++		.set_fmt = wm8753_mode1v_set_dai_fmt,
++		.set_clkdiv = wm8753_set_dai_clkdiv,
++		.set_pll = wm8753_set_dai_pll,
++		.set_sysclk = wm8753_set_dai_sysclk,
++	},
++},
++/* DAI HiFi mode 2 - dummy */
++{	.name = "WM8753 HiFi",
++	.id = 2,
++},
++/* DAI Voice mode 2 */
++{	.name = "WM8753 Voice",
++	.id = 2,
++	.playback = {
++		.stream_name = "Voice Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.ops = {
++		.hw_params = wm8753_pcm_hw_params,},
++	.dai_ops = {
++		.digital_mute = wm8753_mute,
++		.set_fmt = wm8753_mode2_set_dai_fmt,
++		.set_clkdiv = wm8753_set_dai_clkdiv,
++		.set_pll = wm8753_set_dai_pll,
++		.set_sysclk = wm8753_set_dai_sysclk,
++	},
++},
++/* DAI HiFi mode 3 */
++{	.name = "WM8753 HiFi",
++	.id = 3,
++	.playback = {
++		.stream_name = "HiFi Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.ops = {
++		.hw_params = wm8753_i2s_hw_params,},
++	.dai_ops = {
++		.digital_mute = wm8753_mute,
++		.set_fmt = wm8753_mode3_4_set_dai_fmt,
++		.set_clkdiv = wm8753_set_dai_clkdiv,
++		.set_pll = wm8753_set_dai_pll,
++		.set_sysclk = wm8753_set_dai_sysclk,
++	},
++},
++/* DAI Voice mode 3 - dummy */
++{	.name = "WM8753 Voice",
++	.id = 3,
++},
++/* DAI HiFi mode 4 */
++{	.name = "WM8753 HiFi",
++	.id = 4,
++	.playback = {
++		.stream_name = "HiFi Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8753_RATES,
++		.formats = WM8753_FORMATS,},
++	.ops = {
++		.hw_params = wm8753_i2s_hw_params,},
++	.dai_ops = {
++		.digital_mute = wm8753_mute,
++		.set_fmt = wm8753_mode3_4_set_dai_fmt,
++		.set_clkdiv = wm8753_set_dai_clkdiv,
++		.set_pll = wm8753_set_dai_pll,
++		.set_sysclk = wm8753_set_dai_sysclk,
++	},
++},
++/* DAI Voice mode 4 - dummy */
++{	.name = "WM8753 Voice",
++	.id = 4,
++},
++};
++
++struct snd_soc_codec_dai wm8753_dai[2];
++EXPORT_SYMBOL_GPL(wm8753_dai);
++
++static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
++{
++	if (mode < 4) {
++		wm8753_dai[0] = wm8753_all_dai[mode << 1];
++		wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
++	}
++	wm8753_dai[0].codec = codec;
++	wm8753_dai[1].codec = codec;
++}
++
++static void wm8753_work(struct work_struct *work)
++{
++	struct snd_soc_codec *codec =
++		container_of(work, struct snd_soc_codec, delayed_work.work);
++	wm8753_dapm_event(codec, codec->dapm_state);
++}
++
++static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++
++	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int wm8753_resume(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++	int i;
++	u8 data[2];
++	u16 *cache = codec->reg_cache;
++
++	/* Sync reg_cache with the hardware */
++	for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
++		if (i + 1 == WM8753_RESET)
++			continue;
++		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++
++	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++	/* charge wm8753 caps */
++	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
++		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
++		codec->dapm_state = SNDRV_CTL_POWER_D0;
++		queue_delayed_work(wm8753_workq, &codec->delayed_work,
++			msecs_to_jiffies(caps_charge));
++	}
++
++	return 0;
++}
++
++/*
++ * initialise the WM8753 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8753_init(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int reg, ret = 0;
++
++	codec->name = "WM8753";
++	codec->owner = THIS_MODULE;
++	codec->read = wm8753_read_reg_cache;
++	codec->write = wm8753_write;
++	codec->dapm_event = wm8753_dapm_event;
++	codec->dai = wm8753_dai;
++	codec->num_dai = 2;
++	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
++
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8753_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, wm8753_reg,
++		sizeof(u16) * ARRAY_SIZE(wm8753_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8753_reg);
++	wm8753_set_dai_mode(codec, 0);
++
++	wm8753_reset(codec);
++
++	/* register pcms */
++	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++	if (ret < 0) {
++		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,
++		&codec->delayed_work, msecs_to_jiffies(caps_charge));
++
++	/* set the update bits */
++	reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
++	wm8753_write(codec, WM8753_LDAC, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_RDAC);
++	wm8753_write(codec, WM8753_RDAC, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V);
++	wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V);
++	wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V);
++	wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V);
++	wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_LINVOL);
++	wm8753_write(codec, WM8753_LINVOL, reg | 0x0100);
++	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
++	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
++
++	wm8753_add_controls(codec);
++	wm8753_add_widgets(codec);
++	ret = snd_soc_register_card(socdev);
++	if (ret < 0) {
++		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;
++	struct wm8753_priv *wm8753;
++	int ret = 0;
++
++	info("WM8753 Audio Codec %s", WM8753_VERSION);
++
++	setup = socdev->codec_data;
++	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++	if (codec == NULL)
++		return -ENOMEM;
++
++	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
++	if (wm8753 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = wm8753;
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++	wm8753_socdev = socdev;
++	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
++	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->private_data);
++	kfree(codec);
++
++	return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8753 = {
++	.probe = 	wm8753_probe,
++	.remove = 	wm8753_remove,
++	.suspend = 	wm8753_suspend,
++	.resume =	wm8753_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
++
++MODULE_DESCRIPTION("ASoC WM8753 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,126 @@
++/*
++ * wm8753.h  --  audio driver for WM8753
++ *
++ * Copyright 2003 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ */
++
++#ifndef _WM8753_H
++#define _WM8753_H
++
++/* WM8753 register space */
++
++#define WM8753_DAC		0x01
++#define WM8753_ADC		0x02
++#define WM8753_PCM		0x03
++#define WM8753_HIFI		0x04
++#define WM8753_IOCTL	0x05
++#define WM8753_SRATE1	0x06
++#define WM8753_SRATE2	0x07
++#define WM8753_LDAC		0x08
++#define WM8753_RDAC		0x09
++#define WM8753_BASS		0x0a
++#define WM8753_TREBLE	0x0b
++#define WM8753_ALC1		0x0c
++#define WM8753_ALC2		0x0d
++#define WM8753_ALC3		0x0e
++#define WM8753_NGATE	0x0f
++#define WM8753_LADC		0x10
++#define WM8753_RADC		0x11
++#define WM8753_ADCTL1	0x12
++#define WM8753_3D		0x13
++#define WM8753_PWR1		0x14
++#define WM8753_PWR2		0x15
++#define WM8753_PWR3		0x16
++#define WM8753_PWR4		0x17
++#define WM8753_ID		0x18
++#define WM8753_INTPOL	0x19
++#define WM8753_INTEN	0x1a
++#define WM8753_GPIO1	0x1b
++#define WM8753_GPIO2	0x1c
++#define WM8753_RESET	0x1f
++#define WM8753_RECMIX1	0x20
++#define WM8753_RECMIX2	0x21
++#define WM8753_LOUTM1	0x22
++#define WM8753_LOUTM2	0x23
++#define WM8753_ROUTM1	0x24
++#define WM8753_ROUTM2	0x25
++#define WM8753_MOUTM1	0x26
++#define WM8753_MOUTM2	0x27
++#define WM8753_LOUT1V	0x28
++#define WM8753_ROUT1V	0x29
++#define WM8753_LOUT2V	0x2a
++#define WM8753_ROUT2V	0x2b
++#define WM8753_MOUTV	0x2c
++#define WM8753_OUTCTL	0x2d
++#define WM8753_ADCIN	0x2e
++#define WM8753_INCTL1	0x2f
++#define WM8753_INCTL2	0x30
++#define WM8753_LINVOL	0x31
++#define WM8753_RINVOL	0x32
++#define WM8753_MICBIAS	0x33
++#define WM8753_CLOCK	0x34
++#define WM8753_PLL1CTL1	0x35
++#define WM8753_PLL1CTL2	0x36
++#define WM8753_PLL1CTL3	0x37
++#define WM8753_PLL1CTL4	0x38
++#define WM8753_PLL2CTL1	0x39
++#define WM8753_PLL2CTL2	0x3a
++#define WM8753_PLL2CTL3	0x3b
++#define WM8753_PLL2CTL4	0x3c
++#define WM8753_BIASCTL	0x3d
++#define WM8753_ADCTL2	0x3f
++
++struct wm8753_setup_data {
++	unsigned short i2c_address;
++};
++
++#define WM8753_PLL1			0
++#define WM8753_PLL2			1
++
++/* clock inputs */
++#define WM8753_MCLK			0
++#define WM8753_PCMCLK		1
++
++/* clock divider id's */
++#define WM8753_PCMDIV		0
++#define WM8753_BCLKDIV		1
++#define WM8753_VXCLKDIV		2
++
++/* PCM clock dividers */
++#define WM8753_PCM_DIV_1	(0 << 6)
++#define WM8753_PCM_DIV_3	(2 << 6)
++#define WM8753_PCM_DIV_5_5	(3 << 6)
++#define WM8753_PCM_DIV_2	(4 << 6)
++#define WM8753_PCM_DIV_4	(5 << 6)
++#define WM8753_PCM_DIV_6	(6 << 6)
++#define WM8753_PCM_DIV_8	(7 << 6)
++
++/* BCLK clock dividers */
++#define WM8753_BCLK_DIV_1	(0 << 3)
++#define WM8753_BCLK_DIV_2	(1 << 3)
++#define WM8753_BCLK_DIV_4	(2 << 3)
++#define WM8753_BCLK_DIV_8	(3 << 3)
++#define WM8753_BCLK_DIV_16	(4 << 3)
++
++/* VXCLK clock dividers */
++#define WM8753_VXCLK_DIV_1	(0 << 6)
++#define WM8753_VXCLK_DIV_2	(1 << 6)
++#define WM8753_VXCLK_DIV_4	(2 << 6)
++#define WM8753_VXCLK_DIV_8	(3 << 6)
++#define WM8753_VXCLK_DIV_16	(4 << 6)
++
++#define WM8753_DAI_HIFI		0
++#define WM8753_DAI_VOICE	1
++
++extern struct snd_soc_codec_dai wm8753_dai[2];
++extern struct snd_soc_codec_device soc_codec_dev_wm8753;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,596 @@
++/*
++ * wm8772.c  --  WM8772 ALSA Soc Audio driver
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8772.h"
++
++#define AUDIO_NAME "WM8772"
++#define WM8772_VERSION "0.4"
++
++/* codec private data */
++struct wm8772_priv {
++	unsigned int adcclk;
++	unsigned int dacclk;
++};
++
++/*
++ * wm8772 register cache
++ * We can't read the WM8772 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8772_reg[] = {
++	0x00ff, 0x00ff, 0x0120, 0x0000,  /*  0 */
++	0x00ff, 0x00ff, 0x00ff, 0x00ff,  /*  4 */
++	0x00ff, 0x0000, 0x0080, 0x0040,  /*  8 */
++	0x0000
++};
++
++/*
++ * read wm8772 register cache
++ */
++static inline unsigned int wm8772_read_reg_cache(struct snd_soc_codec * codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg > WM8772_CACHE_REGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write wm8772 register cache
++ */
++static inline void wm8772_write_reg_cache(struct snd_soc_codec * codec,
++	unsigned int reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg > WM8772_CACHE_REGNUM)
++		return;
++	cache[reg] = value;
++}
++
++static int wm8772_write(struct snd_soc_codec * codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D9 WM8772 register offset
++	 *   D8...D0 register data
++	 */
++	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++	data[1] = value & 0x00ff;
++
++	wm8772_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -1;
++}
++
++#define wm8772_reset(c)	wm8772_write(c, WM8772_RESET, 0)
++
++/*
++ * WM8772 Controls
++ */
++static const char *wm8772_zero_flag[] = {"All Ch", "Ch 1", "Ch 2", "Ch3"};
++
++static const struct soc_enum wm8772_enum[] = {
++SOC_ENUM_SINGLE(WM8772_DACCTRL, 0, 4, wm8772_zero_flag),
++};
++
++static const struct snd_kcontrol_new wm8772_snd_controls[] = {
++
++SOC_SINGLE("Left1 Playback Volume", WM8772_LDAC1VOL, 0, 255, 0),
++SOC_SINGLE("Left2 Playback Volume", WM8772_LDAC2VOL, 0, 255, 0),
++SOC_SINGLE("Left3 Playback Volume", WM8772_LDAC3VOL, 0, 255, 0),
++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC1VOL, 0, 255, 0),
++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC2VOL, 0, 255, 0),
++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC3VOL, 0, 255, 0),
++SOC_SINGLE("Master Playback Volume", WM8772_MDACVOL, 0, 255, 0),
++
++SOC_SINGLE("Playback Switch", WM8772_DACCH, 0, 1, 0),
++SOC_SINGLE("Capture Switch", WM8772_ADCCTRL, 2, 1, 0),
++
++SOC_SINGLE("Demp1 Playback Switch", WM8772_DACCTRL, 6, 1, 0),
++SOC_SINGLE("Demp2 Playback Switch", WM8772_DACCTRL, 7, 1, 0),
++SOC_SINGLE("Demp3 Playback Switch", WM8772_DACCTRL, 8, 1, 0),
++
++SOC_SINGLE("Phase Invert 1 Switch", WM8772_IFACE, 6, 1, 0),
++SOC_SINGLE("Phase Invert 2 Switch", WM8772_IFACE, 7, 1, 0),
++SOC_SINGLE("Phase Invert 3 Switch", WM8772_IFACE, 8, 1, 0),
++
++SOC_SINGLE("Playback ZC Switch", WM8772_DACCTRL, 0, 1, 0),
++
++SOC_SINGLE("Capture High Pass Switch", WM8772_ADCCTRL, 3, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm8772_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8772_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8772_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++	return 0;
++}
++
++static int wm8772_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct wm8772_priv *wm8772 = codec->private_data;
++
++	switch (freq) {
++	case 4096000:
++	case 5644800:
++	case 6144000:
++	case 8192000:
++	case 8467000:
++	case 9216000:
++	case 11289600:
++	case 12000000:
++	case 12288000:
++	case 16934400:
++	case 18432000:
++	case 22579200:
++	case 24576000:
++	case 33868800:
++	case 36864000:
++		if (clk_id == WM8772_DACCLK) {
++			wm8772->dacclk = freq;
++			return 0;
++		} else if (clk_id == WM8772_ADCCLK) {
++			wm8772->adcclk = freq;
++			return 0;
++		}
++	}
++	return -EINVAL;
++}
++
++static int wm8772_set_dac_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0x1f0;
++	u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0x1ef;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		diface_ctrl |= 0x0010;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		diface |= 0x0002;
++		break;
++	case SND_SOC_DAIFMT_RIGHT_J:
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		diface |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_DSP_A:
++		diface |= 0x0003;
++		break;
++	case SND_SOC_DAIFMT_DSP_B:
++		diface |= 0x0007;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
++	case SND_SOC_DAIFMT_NB_NF:
++		break;
++	case SND_SOC_DAIFMT_IB_NF:
++		diface |= 0x0008;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	wm8772_write(codec, WM8772_DACRATE, diface_ctrl);
++	wm8772_write(codec, WM8772_IFACE, diface);
++	return 0;
++}
++
++static int wm8772_set_adc_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 aiface = 0;
++	u16 aiface_ctrl = wm8772_read_reg_cache(codec, WM8772_ADCCTRL) & 0x1cf;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		aiface |= 0x0010;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		aiface |= 0x0002;
++		break;
++	case SND_SOC_DAIFMT_RIGHT_J:
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		aiface |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_DSP_A:
++		aiface |= 0x0003;
++		break;
++	case SND_SOC_DAIFMT_DSP_B:
++		aiface |= 0x0003;
++		aiface_ctrl |= 0x0010;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
++	case SND_SOC_DAIFMT_NB_NF:
++		break;
++	case SND_SOC_DAIFMT_IB_NF:
++		aiface_ctrl |= 0x0020;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	wm8772_write(codec, WM8772_ADCCTRL, aiface_ctrl);
++	wm8772_write(codec, WM8772_ADCRATE, aiface);
++	return 0;
++}
++
++static int wm8772_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8772_priv *wm8772 = codec->private_data;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++
++		u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0x1cf;
++		u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0x3f;
++
++		/* bit size */
++		switch (params_format(params)) {
++		case SNDRV_PCM_FORMAT_S16_LE:
++			break;
++		case SNDRV_PCM_FORMAT_S20_3LE:
++			diface |= 0x0010;
++			break;
++		case SNDRV_PCM_FORMAT_S24_3LE:
++			diface |= 0x0020;
++			break;
++		case SNDRV_PCM_FORMAT_S32_LE:
++			diface |= 0x0030;
++			break;
++		}
++
++		/* set rate */
++		switch (wm8772->dacclk / params_rate(params)) {
++		case 768:
++			diface_ctrl |= (0x5 << 6);
++			break;
++		case 512:
++			diface_ctrl |= (0x4 << 6);
++			break;
++		case 384:
++			diface_ctrl |= (0x3 << 6);
++			break;
++		case 256:
++			diface_ctrl |= (0x2 << 6);
++			break;
++		case 192:
++			diface_ctrl |= (0x1 << 6);
++			break;
++		}
++
++		wm8772_write(codec, WM8772_DACRATE, diface_ctrl);
++		wm8772_write(codec, WM8772_IFACE, diface);
++
++	} else {
++
++		u16 aiface = wm8772_read_reg_cache(codec, WM8772_ADCRATE) & 0x113;
++
++		/* bit size */
++		switch (params_format(params)) {
++		case SNDRV_PCM_FORMAT_S16_LE:
++			break;
++		case SNDRV_PCM_FORMAT_S20_3LE:
++			aiface |= 0x0004;
++			break;
++		case SNDRV_PCM_FORMAT_S24_LE:
++			aiface |= 0x0008;
++			break;
++		case SNDRV_PCM_FORMAT_S32_LE:
++			aiface |= 0x000c;
++			break;
++		}
++
++		/* set rate */
++		switch (wm8772->adcclk / params_rate(params)) {
++		case 768:
++			aiface |= (0x5 << 5);
++			break;
++		case 512:
++			aiface |= (0x4 << 5);
++			break;
++		case 384:
++			aiface |= (0x3 << 5);
++			break;
++		case 256:
++			aiface |= (0x2 << 5);
++			break;
++		}
++
++		wm8772_write(codec, WM8772_ADCRATE, aiface);
++	}
++
++	return 0;
++}
++
++static int wm8772_dapm_event(struct snd_soc_codec *codec, int event)
++{
++	u16 master = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xffe0;
++
++	switch (event) {
++		case SNDRV_CTL_POWER_D0: /* full On */
++			/* vref/mid, clk and osc on, dac unmute, active */
++			wm8772_write(codec, WM8772_DACRATE, master);
++			break;
++		case SNDRV_CTL_POWER_D1: /* partial On */
++		case SNDRV_CTL_POWER_D2: /* partial On */
++			break;
++		case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++			/* everything off except vref/vmid, dac mute, inactive */
++			wm8772_write(codec, WM8772_DACRATE, master | 0x0f);
++			break;
++		case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++			/* everything off, dac mute, inactive */
++			wm8772_write(codec, WM8772_DACRATE, master | 0x1f);
++			break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++struct snd_soc_codec_dai wm8772_dai[] = {
++{
++	.name = "WM8772",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 2,
++		.channels_max = 6,
++	},
++	.ops = {
++		.hw_params = wm8772_hw_params,
++	},
++	.dai_ops = {
++		.set_fmt = wm8772_set_dac_dai_fmt,
++		.set_sysclk = wm8772_set_dai_sysclk,
++	},
++},
++{
++	.name = "WM8772",
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 2,
++		.channels_max = 2,
++	},
++	.ops = {
++		.hw_params = wm8772_hw_params,
++	},
++	.dai_ops = {
++		.set_fmt = wm8772_set_adc_dai_fmt,
++		.set_sysclk = wm8772_set_dai_sysclk,
++	},
++},
++};
++EXPORT_SYMBOL_GPL(wm8772_dai);
++
++static int wm8772_suspend(struct platform_device *pdev, pm_message_t state)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++
++	wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int wm8772_resume(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++	int i;
++	u8 data[2];
++	u16 *cache = codec->reg_cache;
++
++	/* Sync reg_cache with the hardware */
++	for (i = 0; i < ARRAY_SIZE(wm8772_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8772_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the WM8772 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8772_init(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int reg, ret = 0;
++
++	codec->name = "WM8772";
++	codec->owner = THIS_MODULE;
++	codec->read = wm8772_read_reg_cache;
++	codec->write = wm8772_write;
++	codec->dapm_event = wm8772_dapm_event;
++	codec->dai = wm8772_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(wm8772_reg);
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8772_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, wm8772_reg,
++		sizeof(u16) * ARRAY_SIZE(wm8772_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8772_reg);
++
++	wm8772_reset(codec);
++
++	/* register pcms */
++	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++	if(ret < 0) {
++		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;
++	struct wm8772_priv *wm8772;
++	int ret = 0;
++
++	printk(KERN_INFO "WM8772 Audio Codec %s", WM8772_VERSION);
++
++	setup = socdev->codec_data;
++	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++	if (codec == NULL)
++		return -ENOMEM;
++
++	wm8772 = kzalloc(sizeof(struct wm8772_priv), GFP_KERNEL);
++	if (wm8772 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = wm8772;
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++
++	wm8772_socdev = socdev;
++
++	/* Add other interfaces here */
++#warning do SPI device probe here and then call wm8772_init()
++
++	return ret;
++}
++
++/* power down chip */
++static int wm8772_remove(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_codec *codec = socdev->codec;
++
++	if (codec->control_data)
++		wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++
++	snd_soc_free_pcms(socdev);
++	kfree(codec->private_data);
++	kfree(codec);
++
++	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-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,46 @@
++/*
++ * wm8772.h  --  audio driver for WM8772
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ */
++
++#ifndef _WM8772_H
++#define _WM8772_H
++
++/* WM8772 register space */
++
++#define WM8772_LDAC1VOL   0x00
++#define WM8772_RDAC1VOL   0x01
++#define WM8772_DACCH      0x02
++#define WM8772_IFACE      0x03
++#define WM8772_LDAC2VOL   0x04
++#define WM8772_RDAC2VOL   0x05
++#define WM8772_LDAC3VOL   0x06
++#define WM8772_RDAC3VOL   0x07
++#define WM8772_MDACVOL    0x08
++#define WM8772_DACCTRL    0x09
++#define WM8772_DACRATE    0x0a
++#define WM8772_ADCRATE    0x0b
++#define WM8772_ADCCTRL    0x0c
++#define WM8772_RESET	  0x1f
++
++#define WM8772_CACHE_REGNUM 	10
++
++#define WM8772_DACCLK	0
++#define WM8772_ADCCLK	1
++
++#define WM8753_DAI_DAC		0
++#define WM8753_DAI_ADC		1
++
++extern struct snd_soc_codec_dai wm8772_dai[2];
++extern struct snd_soc_codec_device soc_codec_dev_wm8772;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,965 @@
++/*
++ * wm8971.c  --  WM8971 ALSA SoC Audio driver
++ *
++ * Copyright 2005 Lab126, Inc.
++ *
++ * Author: Kenneth Kiraly <kiraly at lab126.com>
++ *
++ * Based on wm8753.c by Liam Girdwood
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "wm8971.h"
++
++#define AUDIO_NAME "wm8971"
++#define WM8971_VERSION "0.9"
++
++#undef	WM8971_DEBUG
++
++#ifdef WM8971_DEBUG
++#define dbg(format, arg...) \
++	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++#define	WM8971_REG_COUNT		43
++
++static struct workqueue_struct *wm8971_workq = NULL;
++
++/* codec private data */
++struct wm8971_priv {
++	unsigned int sysclk;
++};
++
++/*
++ * wm8971 register cache
++ * We can't read the WM8971 register space when we
++ * are using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8971_reg[] = {
++	0x0097, 0x0097, 0x0079, 0x0079,  /*  0 */
++	0x0000, 0x0008, 0x0000, 0x000a,  /*  4 */
++	0x0000, 0x0000, 0x00ff, 0x00ff,  /*  8 */
++	0x000f, 0x000f, 0x0000, 0x0000,  /* 12 */
++	0x0000, 0x007b, 0x0000, 0x0032,  /* 16 */
++	0x0000, 0x00c3, 0x00c3, 0x00c0,  /* 20 */
++	0x0000, 0x0000, 0x0000, 0x0000,  /* 24 */
++	0x0000, 0x0000, 0x0000, 0x0000,  /* 28 */
++	0x0000, 0x0000, 0x0050, 0x0050,  /* 32 */
++	0x0050, 0x0050, 0x0050, 0x0050,  /* 36 */
++	0x0079, 0x0079, 0x0079,          /* 40 */
++};
++
++static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg < WM8971_REG_COUNT)
++		return cache[reg];
++
++	return -1;
++}
++
++static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg < WM8971_REG_COUNT)
++		cache[reg] = value;
++}
++
++static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D9 WM8753 register offset
++	 *   D8...D0 register data
++	 */
++	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++	data[1] = value & 0x00ff;
++
++	wm8971_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -EIO;
++}
++
++#define wm8971_reset(c)	wm8971_write(c, WM8971_RESET, 0)
++
++/* WM8971 Controls */
++static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" };
++static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz",
++	"200Hz @ 48kHz" };
++static const char *wm8971_treble[] = { "8kHz", "4kHz" };
++static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" };
++static const char *wm8971_ng_type[] = { "Constant PGA Gain",
++	"Mute ADC Output" };
++static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
++static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)",
++	"Mono (Right)", "Digital Mono"};
++static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" };
++static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA",
++	"Differential"};
++static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA",
++	"Differential"};
++static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"};
++static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"};
++static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert",
++	"L + R Invert"};
++
++static const struct soc_enum wm8971_enum[] = {
++	SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass),			/* 0 */
++	SOC_ENUM_SINGLE(WM8971_BASS, 6, 2, wm8971_bass_filter),
++	SOC_ENUM_SINGLE(WM8971_TREBLE, 6, 2, wm8971_treble),
++	SOC_ENUM_SINGLE(WM8971_ALC1, 7, 4, wm8971_alc_func),
++	SOC_ENUM_SINGLE(WM8971_NGATE, 1, 2, wm8971_ng_type),        /* 4 */
++	SOC_ENUM_SINGLE(WM8971_ADCDAC, 1, 4, wm8971_deemp),
++	SOC_ENUM_SINGLE(WM8971_ADCTL1, 4, 4, wm8971_mono_mux),
++	SOC_ENUM_SINGLE(WM8971_ADCTL1, 1, 2, wm8971_dac_phase),
++	SOC_ENUM_SINGLE(WM8971_LOUTM1, 0, 5, wm8971_lline_mux),     /* 8 */
++	SOC_ENUM_SINGLE(WM8971_ROUTM1, 0, 5, wm8971_rline_mux),
++	SOC_ENUM_SINGLE(WM8971_LADCIN, 6, 4, wm8971_lpga_sel),
++	SOC_ENUM_SINGLE(WM8971_RADCIN, 6, 4, wm8971_rpga_sel),
++	SOC_ENUM_SINGLE(WM8971_ADCDAC, 5, 4, wm8971_adcpol),        /* 12 */
++	SOC_ENUM_SINGLE(WM8971_ADCIN, 6, 4, wm8971_mono_mux),
++};
++
++static const struct snd_kcontrol_new wm8971_snd_controls[] = {
++	SOC_DOUBLE_R("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, 0, 63, 0),
++	SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL, 6, 1, 0),
++	SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1),
++
++	SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V,
++		WM8971_ROUT1V, 7, 1, 0),
++	SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V,
++		WM8971_ROUT2V, 7, 1, 0),
++	SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0),
++
++	SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0),
++
++	SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1,
++		WM8971_LOUTM2, 4, 7, 1),
++	SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1,
++		WM8971_ROUTM2, 4, 7, 1),
++	SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1,
++		WM8971_MOUTM2, 4, 7, 1),
++
++	SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V,
++		WM8971_ROUT1V, 0, 127, 0),
++	SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V,
++		WM8971_ROUT2V, 0, 127, 0),
++
++	SOC_ENUM("Bass Boost", wm8971_enum[0]),
++	SOC_ENUM("Bass Filter", wm8971_enum[1]),
++	SOC_SINGLE("Bass Volume", WM8971_BASS, 0, 7, 1),
++
++	SOC_SINGLE("Treble Volume", WM8971_TREBLE, 0, 7, 0),
++	SOC_ENUM("Treble Cut-off", wm8971_enum[2]),
++
++	SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1),
++
++	SOC_SINGLE("ALC Target Volume", WM8971_ALC1, 0, 7, 0),
++	SOC_SINGLE("ALC Max Volume", WM8971_ALC1, 4, 7, 0),
++
++	SOC_SINGLE("ALC Capture Target Volume", WM8971_ALC1, 0, 7, 0),
++	SOC_SINGLE("ALC Capture Max Volume", WM8971_ALC1, 4, 7, 0),
++	SOC_ENUM("ALC Capture Function", wm8971_enum[3]),
++	SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0),
++	SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0),
++	SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0),
++	SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0),
++	SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0),
++	SOC_ENUM("ALC Capture NG Type", wm8971_enum[4]),
++	SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0),
++
++	SOC_SINGLE("Capture 6dB Attenuate", WM8971_ADCDAC, 8, 1, 0),
++	SOC_SINGLE("Playback 6dB Attenuate", WM8971_ADCDAC, 7, 1, 0),
++
++    SOC_ENUM("Playback De-emphasis", wm8971_enum[5]),
++	SOC_ENUM("Playback Function", wm8971_enum[6]),
++	SOC_ENUM("Playback Phase", wm8971_enum[7]),
++
++	SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
++};
++
++/* add non-DAPM controls */
++static int wm8971_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8971_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++	return 0;
++}
++
++/*
++ * DAPM Controls
++ */
++
++/* Left Mixer */
++static const struct snd_kcontrol_new wm8971_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Playback Switch", WM8971_LOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_LOUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8971_LOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_LOUTM2, 7, 1, 0),
++};
++
++/* Right Mixer */
++static const struct snd_kcontrol_new wm8971_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8971_ROUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_ROUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Playback Switch", WM8971_ROUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_ROUTM2, 7, 1, 0),
++};
++
++/* Mono Mixer */
++static const struct snd_kcontrol_new wm8971_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left Playback Switch", WM8971_MOUTM1, 8, 1, 0),
++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_MOUTM1, 7, 1, 0),
++SOC_DAPM_SINGLE("Right Playback Switch", WM8971_MOUTM2, 8, 1, 0),
++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0),
++};
++
++/* Left Line Mux */
++static const struct snd_kcontrol_new wm8971_left_line_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[8]);
++
++/* Right Line Mux */
++static const struct snd_kcontrol_new wm8971_right_line_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[9]);
++
++/* Left PGA Mux */
++static const struct snd_kcontrol_new wm8971_left_pga_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[10]);
++
++/* Right PGA Mux */
++static const struct snd_kcontrol_new wm8971_right_pga_controls =
++SOC_DAPM_ENUM("Route", wm8971_enum[11]);
++
++/* Mono ADC Mux */
++static const struct snd_kcontrol_new wm8971_monomux_controls =
++SOC_DAPM_ENUM("Route", 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;
++}
++
++static int wm8971_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct wm8971_priv *wm8971 = codec->private_data;
++
++	switch (freq) {
++	case 11289600:
++	case 12000000:
++	case 12288000:
++	case 16934400:
++	case 18432000:
++		wm8971->sysclk = freq;
++		return 0;
++	}
++	return -EINVAL;
++}
++
++static int wm8971_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = 0;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		iface = 0x0040;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	wm8971_write(codec, WM8971_IFACE, iface);
++	return 0;
++}
++
++static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8971_priv *wm8971 = codec->private_data;
++	u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
++	u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
++	int coeff = get_coeff(wm8971->sysclk, params_rate(params));
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0008;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		iface |= 0x000c;
++		break;
++	}
++
++	/* set iface & srate */
++	wm8971_write(codec, WM8971_IFACE, iface);
++	if (coeff >= 0)
++		wm8971_write(codec, WM8971_SRATE, srate |
++			(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
++
++	return 0;
++}
++
++static int wm8971_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	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;
++}
++
++#define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_codec_dai wm8971_dai = {
++	.name = "WM8971",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8971_RATES,
++		.formats = WM8971_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8971_RATES,
++		.formats = WM8971_FORMATS,},
++	.ops = {
++		.hw_params = wm8971_pcm_hw_params,
++	},
++	.dai_ops = {
++		.digital_mute = wm8971_mute,
++		.set_fmt = wm8971_set_dai_fmt,
++		.set_sysclk = wm8971_set_dai_sysclk,
++	},
++};
++EXPORT_SYMBOL_GPL(wm8971_dai);
++
++static void wm8971_work(struct work_struct *work)
++{
++	struct snd_soc_codec *codec =
++		container_of(work, struct snd_soc_codec, delayed_work.work);
++	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, &codec->delayed_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, &codec->delayed_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;
++	struct wm8971_priv *wm8971;
++	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;
++
++	wm8971 = kzalloc(sizeof(struct wm8971_priv), GFP_KERNEL);
++	if (wm8971 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = wm8971;
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++	wm8971_socdev = socdev;
++
++	INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work);
++	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->private_data);
++	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-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,63 @@
++/*
++ * wm8971.h  --  audio driver for WM8971
++ *
++ * Copyright 2005 Lab126, Inc.
++ *
++ * Author: Kenneth Kiraly <kiraly at 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_IFACE	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
++
++#define WM8971_SYSCLK	0
++
++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-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,867 @@
++/*
++ * wm8974.c  --  WM8974 ALSA Soc Audio driver
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ *
++ * Author: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License 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 "0.6"
++
++/*
++ * 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,
++};
++
++/*
++ * 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 wm8974_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	int i;
++	u16 reg;
++
++	if(freq_in == 0 || freq_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 (freq_in == pll[i].in_hz && freq_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;
++}
++
++/*
++ * Configure WM8974 clock dividers.
++ */
++static int wm8974_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	switch (div_id) {
++	case WM8974_OPCLKDIV:
++		reg = wm8974_read_reg_cache(codec, WM8974_GPIO & 0x1cf);
++		wm8974_write(codec, WM8974_GPIO, reg | div);
++		break;
++	case WM8974_MCLKDIV:
++		reg = wm8974_read_reg_cache(codec, WM8974_CLOCK & 0x1f);
++		wm8974_write(codec, WM8974_CLOCK, reg | div);
++		break;
++	case WM8974_ADCCLK:
++		reg = wm8974_read_reg_cache(codec, WM8974_ADC & 0x1f7);
++		wm8974_write(codec, WM8974_ADC, reg | div);
++		break;
++	case WM8974_DACCLK:
++		reg = wm8974_read_reg_cache(codec, WM8974_DAC & 0x1f7);
++		wm8974_write(codec, WM8974_DAC, reg | div);
++		break;
++	case WM8974_BCLKDIV:
++		reg = wm8974_read_reg_cache(codec, WM8974_CLOCK & 0x1e3);
++		wm8974_write(codec, WM8974_CLOCK, reg | div);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static int wm8974_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = 0;
++	u16 clk = wm8974_read_reg_cache(codec, WM8974_CLOCK) & 0x1fe;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		clk |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	wm8974_write(codec, WM8974_IFACE, iface);
++	wm8974_write(codec, WM8974_CLOCK, clk);
++	return 0;
++}
++
++static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 iface = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x19f;
++	u16 adn = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x1f1;
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0020;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0040;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		iface |= 0x0060;
++		break;
++	}
++
++	/* filter coefficient */
++	switch (params_rate(params)) {
++	case SNDRV_PCM_RATE_8000:
++		adn |= 0x5 << 1;
++		break;
++	case SNDRV_PCM_RATE_11025:
++		adn |= 0x4 << 1;
++		break;
++	case SNDRV_PCM_RATE_16000:
++		adn |= 0x3 << 1;
++		break;
++	case SNDRV_PCM_RATE_22050:
++		adn |= 0x2 << 1;
++		break;
++	case SNDRV_PCM_RATE_32000:
++		adn |= 0x1 << 1;
++		break;
++	case SNDRV_PCM_RATE_44100:
++		break;
++	}
++
++	wm8974_write(codec, WM8974_IFACE, iface);
++	wm8974_write(codec, WM8974_ADD, adn);
++	return 0;
++}
++
++static int wm8974_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	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;
++}
++
++#define WM8974_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)
++
++#define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_codec_dai wm8974_dai = {
++	.name = "WM8974 HiFi",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8974_RATES,
++		.formats = WM8974_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8974_RATES,
++		.formats = WM8974_FORMATS,},
++	.ops = {
++		.hw_params = wm8974_pcm_hw_params,
++	},
++	.dai_ops = {
++		.digital_mute = wm8974_mute,
++		.set_fmt = wm8974_set_dai_fmt,
++		.set_clkdiv = wm8974_set_dai_clkdiv,
++		.set_pll = wm8974_set_dai_pll,
++	},
++};
++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-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,104 @@
++/*
++ * 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
++
++/* Clock divider Id's */
++#define WM8974_OPCLKDIV		0
++#define WM8974_MCLKDIV		1
++#define WM8974_ADCCLK		2
++#define WM8974_DACCLK		3
++#define WM8974_BCLKDIV		4
++
++/* DAC clock dividers */
++#define WM8974_DACCLK_F2	(1 << 3)
++#define WM8974_DACCLK_F4	(0 << 3)
++
++/* ADC clock dividers */
++#define WM8974_ADCCLK_F2	(1 << 3)
++#define WM8974_ADCCLK_F4	(0 << 3)
++
++/* PLL Out dividers */
++#define WM8974_OPCLKDIV_1	(0 << 4)
++#define WM8974_OPCLKDIV_2	(1 << 4)
++#define WM8974_OPCLKDIV_3	(2 << 4)
++#define WM8974_OPCLKDIV_4	(3 << 4)
++
++/* BCLK clock dividers */
++#define WM8974_BCLKDIV_1	(0 << 2)
++#define WM8974_BCLKDIV_2	(1 << 2)
++#define WM8974_BCLKDIV_4	(2 << 2)
++#define WM8974_BCLKDIV_8	(3 << 2)
++#define WM8974_BCLKDIV_16	(4 << 2)
++#define WM8974_BCLKDIV_32	(5 << 2)
++
++/* MCLK clock dividers */
++#define WM8974_MCLKDIV_1	(0 << 5)
++#define WM8974_MCLKDIV_1_5	(1 << 5)
++#define WM8974_MCLKDIV_2	(2 << 5)
++#define WM8974_MCLKDIV_3	(3 << 5)
++#define WM8974_MCLKDIV_4	(4 << 5)
++#define WM8974_MCLKDIV_6	(5 << 5)
++#define WM8974_MCLKDIV_8	(6 << 5)
++#define WM8974_MCLKDIV_12	(7 << 5)
++
++
++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-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,767 @@
++/*
++ * wm9712.c  --  ALSA Soc WM9712 codec support
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  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);
++
++/*
++ * 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);
++}
++
++#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
++
++struct snd_soc_codec_dai wm9712_dai[] = {
++{
++	.name = "AC97 HiFi",
++	.playback = {
++		.stream_name = "HiFi Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM9712_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "HiFi Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM9712_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.prepare = ac97_prepare,},
++},
++{
++	.name = "AC97 Aux",
++	.playback = {
++		.stream_name = "Aux Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM9712_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.prepare = ac97_aux_prepare,},
++}
++};
++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-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.h	2007-01-24 12:19:24.000000000 +0100
+@@ -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-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,1218 @@
++/*
++ * wm9713.c  --  ALSA Soc WM9713 codec support
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  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/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include "wm9713.h"
++
++#define WM9713_VERSION "0.12"
++
++struct wm9713_priv {
++	u32 pll_in; /* PLL input frequency */
++	u32 pll_out; /* PLL output 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);
++
++/*
++ * 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, 31, 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 */
++};
++
++static int wm9713_set_pll(struct snd_soc_codec *codec,
++	int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	struct wm9713_priv *wm9713 = codec->private_data;
++	int i;
++	u16 reg, reg2;
++
++	/* turn PLL off ? */
++	if (freq_in == 0 || freq_out == 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);
++		wm9713->pll_out = 0;
++		return 0;
++	}
++
++	for (i = 0; i < ARRAY_SIZE(pll); i++) {
++		if (pll[i].in_hz == freq_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);
++	wm9713->pll_out = freq_out;
++	wm9713->pll_in = freq_in;
++
++	/* wait 10ms AC97 link frames for the link to stabilise */
++	schedule_timeout_interruptible(msecs_to_jiffies(10));
++	return 0;
++}
++
++static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
++}
++
++/*
++ * Tristate the PCM DAI lines, tristate can be disabled by calling
++ * wm9713_set_dai_fmt()
++ */
++static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
++	int tristate)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff;
++
++	if (tristate)
++		ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
++
++	return 0;
++}
++
++/*
++ * Configure WM9713 clock dividers.
++ * Voice DAC needs 256 FS
++ */
++static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	switch (div_id) {
++	case WM9713_PCMCLK_DIV:
++		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff;
++		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
++		break;
++	case WM9713_CLKA_MULT:
++		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd;
++		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
++		break;
++	case WM9713_CLKB_MULT:
++		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb;
++		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
++		break;
++	case WM9713_HIFI_DIV:
++		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff;
++		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
++		break;
++	case WM9713_PCMBCLK_DIV:
++		reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff;
++		ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++};
++
++static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffe2;
++	u16 reg = 0x8000;
++
++	/* clock masters */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_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;
++	}
++
++	/* clock inversion */
++	switch (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;
++	}
++
++	/* DAI format */
++	switch (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;
++	}
++
++	ac97_write(codec, AC97_GPIO_CFG, gpio);
++	ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
++	return 0;
++}
++
++static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
++
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		reg |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		reg |= 0x0008;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		reg |= 0x000c;
++		break;
++	}
++
++	/* enable PCM interface in master mode */
++	ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
++	return 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;
++
++    /* 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;
++
++	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);
++}
++
++#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
++
++#define WM9713_PCM_FORMATS \
++	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
++	 SNDRV_PCM_FORMAT_S24_LE)
++
++struct snd_soc_codec_dai wm9713_dai[] = {
++{
++	.name = "AC97 HiFi",
++	.playback = {
++		.stream_name = "HiFi Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM9713_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "HiFi Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM9713_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.prepare = ac97_hifi_prepare,},
++	},
++	{
++	.name = "AC97 Aux",
++	.playback = {
++		.stream_name = "Aux Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM9713_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.prepare = ac97_aux_prepare,},
++	},
++	{
++	.name = "WM9713 Voice",
++	.playback = {
++		.stream_name = "Voice Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM9713_RATES,
++		.formats = WM9713_PCM_FORMATS,},
++	.capture = {
++		.stream_name = "Voice Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM9713_RATES,
++		.formats = WM9713_PCM_FORMATS,},
++	.ops = {
++		.hw_params = wm9713_pcm_hw_params,
++		.shutdown = wm9713_voiceshutdown,},
++	.dai_ops = {
++		.set_clkdiv = wm9713_set_dai_clkdiv,
++		.set_pll = wm9713_set_dai_pll,
++		.set_fmt = wm9713_set_dai_fmt,
++		.set_tristate = wm9713_set_dai_tristate,
++	},
++	},
++};
++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;
++
++	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_priv *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);
++
++	/* do we need to re-start the PLL ? */
++	if (wm9713->pll_out)
++		wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
++
++	/* 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 (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_priv), 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-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,51 @@
++/*
++ * wm9713.h  --  WM9713 Soc Audio driver
++ */
++
++#ifndef _WM9713_H
++#define _WM9713_H
++
++/* clock inputs */
++#define WM9713_CLKA_PIN			0
++#define WM9713_CLKB_PIN			1
++
++/* clock divider ID's */
++#define WM9713_PCMCLK_DIV		0
++#define WM9713_CLKA_MULT		1
++#define WM9713_CLKB_MULT		2
++#define WM9713_HIFI_DIV			3
++#define WM9713_PCMBCLK_DIV		4
++
++/* PCM clk div */
++#define WM9713_PCMDIV(x)	((x - 1) << 8)
++
++/* HiFi Div */
++#define WM9713_HIFIDIV(x)	((x - 1) << 12)
++
++/* MCLK clock mulitipliers */
++#define WM9713_CLKA_X1		(0 << 1)
++#define WM9713_CLKA_X2		(1 << 1)
++#define WM9713_CLKB_X1		(0 << 2)
++#define WM9713_CLKB_X2		(1 << 2)
++
++/* MCLK clock MUX */
++#define WM9713_CLK_MUX_A		(0 << 0)
++#define WM9713_CLK_MUX_B		(1 << 0)
++
++/* Voice DAI BCLK divider */
++#define WM9713_PCMBCLK_DIV_1	(0 << 9)
++#define WM9713_PCMBCLK_DIV_2	(1 << 9)
++#define WM9713_PCMBCLK_DIV_4	(2 << 9)
++#define WM9713_PCMBCLK_DIV_8	(3 << 9)
++#define WM9713_PCMBCLK_DIV_16	(4 << 9)
++
++#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];
++
++int wm9713_reset(struct snd_soc_codec *codec,  int try_warm);
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/Kconfig
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,134 @@
++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 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).
++
++config SND_PXA2xx_SOC_MAGICIAN
++	tristate "SoC Audio support for HTC Magician"
++	depends on SND_PXA2xx_SOC
++	select SND_PXA2xx_SOC_I2S
++	select SND_PXA2xx_SOC_SSP
++	help
++	  Say Y if you want to add support for SoC audio on the
++	  HTC Magician.
++
++endmenu
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/Makefile
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/Makefile	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,37 @@
++# 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-$(CONFIG_SND_PXA2xx_SOC_SSP) += snd-soc-pxa2xx-ssp.o
++
++# PXA Machine Support
++snd-soc-corgi-objs := corgi.o
++snd-soc-mainstone-wm8731-objs := mainstone_wm8731.o
++snd-soc-mainstone-wm8753-objs := mainstone_wm8753.o
++snd-soc-mainstone-wm8974-objs := mainstone_wm8974.o
++snd-soc-mainstone-wm9713-objs := mainstone_wm9713.o
++snd-soc-mainstone-wm9712-objs := mainstone_wm9712.o
++snd-soc-mainstone-baseband-objs := mainstone_baseband.o
++snd-soc-mainstone-bluetooth-objs := mainstone_bluetooth.o
++snd-soc-poodle-objs := poodle.o
++snd-soc-tosa-objs := tosa.o
++snd-soc-spitz-objs := spitz.o
++snd-soc-magician-objs := magician.o
++
++obj-$(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
++obj-$(CONFIG_SND_PXA2xx_SOC_MAGICIAN) += snd-soc-magician.o
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/corgi.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/corgi.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,383 @@
++/*
++ * corgi.c  --  SoC audio for Corgi
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *          Richard Purdie <richard at 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"
++#include "pxa2xx-i2s.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 int corgi_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int clk = 0;
++	int ret = 0;
++
++	switch (params_rate(params)) {
++	case 8000:
++	case 16000:
++	case 48000:
++	case 96000:
++		clk = 12288000;
++		break;
++	case 11025:
++	case 22050:
++	case 44100:
++		clk = 11289600;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set the I2S system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static struct snd_soc_ops corgi_ops = {
++	.startup = corgi_startup,
++	.hw_params = corgi_hw_params,
++	.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;
++}
++
++/* 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,
++	.ops = &corgi_ops,
++};
++
++/* corgi audio machine driver */
++static struct snd_soc_machine snd_soc_machine_corgi = {
++	.name = "Corgi",
++	.dai_link = &corgi_dai,
++	.num_links = 1,
++};
++
++/* 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-2.6.17.14-fic4.test/sound/soc/pxa/mainstone.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,127 @@
++/*
++ * mainstone.c  --  SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at 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"
++#include "pxa2xx-ac97.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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_baseband.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_baseband.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,212 @@
++/*
++ * mainstone_baseband.c
++ * Mainstone Example Baseband modem  --  ALSA Soc Audio Layer
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  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"
++#include "pxa2xx-ac97.h"
++#include "pxa2xx-ssp.h"
++
++static struct snd_soc_machine mainstone;
++
++/* Do specific baseband PCM voice startup here */
++static int baseband_startup(struct snd_pcm_substream *substream)
++{
++	return 0;
++}
++
++/* Do specific baseband PCM voice shutdown here */
++static void baseband_shutdown (struct snd_pcm_substream *substream)
++{
++}
++
++/* Do specific baseband modem PCM voice hw params init here */
++static int 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 baseband_hw_free(struct snd_pcm_substream *substream)
++{
++	return 0;
++}
++
++/*
++ * Baseband Processor DAI
++ */
++static struct snd_soc_cpu_dai baseband_dai =
++{	.name = "Baseband",
++	.id = 0,
++	.type = SND_SOC_DAI_PCM,
++	.playback = {
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = SNDRV_PCM_RATE_8000,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = SNDRV_PCM_RATE_8000,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.startup = baseband_startup,
++		.shutdown = baseband_shutdown,
++		.hw_params = baseband_hw_params,
++		.hw_free = baseband_hw_free,
++		},
++};
++
++/* 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;
++}
++
++/* 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 = &baseband_dai,
++	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
++},
++};
++
++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_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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("Mainstone Example Baseband PCM Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_bluetooth.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_bluetooth.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,371 @@
++/*
++ * mainstone_bluetooth.c
++ * Mainstone Example Bluetooth  --  ALSA Soc Audio Layer
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  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"
++#include "pxa2xx-i2s.h"
++#include "pxa2xx-ssp.h"
++
++static struct snd_soc_machine mainstone;
++
++/* Do specific bluetooth PCM startup here */
++static int bt_startup(struct snd_pcm_substream *substream)
++{
++	return 0;
++}
++
++/* Do specific bluetooth PCM shutdown here */
++static void bt_shutdown (struct snd_pcm_substream *substream)
++{
++}
++
++/* Do pecific bluetooth PCM hw params init here */
++static int 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 bt_hw_free(struct snd_pcm_substream *substream)
++{
++	return 0;
++}
++
++#define BT_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100)
++
++/*
++ * BT Codec DAI
++ */
++static struct snd_soc_cpu_dai bt_dai =
++{	.name = "Bluetooth",
++	.id = 0,
++	.type = SND_SOC_DAI_PCM,
++	.playback = {
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = BT_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = BT_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.startup = bt_startup,
++		.shutdown = bt_shutdown,
++		.hw_params = bt_hw_params,
++		.hw_free = bt_hw_free,
++		},
++};
++
++/* 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);
++
++	if (machine_spk_func == ucontrol->value.integer.value[0])
++		return 0;
++
++	machine_spk_func = ucontrol->value.integer.value[0];
++	snd_soc_dapm_set_endpoint(codec, "Spk", machine_spk_func);
++	return 1;
++}
++
++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);
++
++	if (machine_spk_func == ucontrol->value.integer.value[0])
++		return 0;
++
++	machine_spk_func = ucontrol->value.integer.value[0];
++	snd_soc_dapm_set_endpoint(codec, "Mic", machine_mic_func);
++	return 1;
++}
++
++/* 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;
++}
++
++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,
++},
++{ /* Voice via BT */
++	.name = "Bluetooth",
++	.stream_name = "Voice",
++	.cpu_dai = &bt_dai,
++	.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
++},
++};
++
++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_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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("Mainstone Example Bluetooth PCM Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8731.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8731.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,203 @@
++/*
++ * mainstone.c  --  SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at 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"
++#include "pxa2xx-i2s.h"
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int clk = 0;
++	int ret = 0;
++
++	switch (params_rate(params)) {
++	case 8000:
++	case 16000:
++	case 48000:
++	case 96000:
++		clk = 12288000;
++		break;
++	case 11025:
++	case 22050:
++	case 44100:
++		clk = 11289600;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set the I2S system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static struct snd_soc_ops mainstone_ops = {
++	.hw_params = mainstone_hw_params,
++};
++
++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;
++}
++
++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,
++	.ops = &mainstone_ops,
++	},
++};
++
++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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM8731 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8753.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8753.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,547 @@
++/*
++ * mainstone.c  --  SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at 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/hardware.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"
++#include "pxa2xx-i2s.h"
++#include "pxa2xx-ssp.h"
++
++/*
++ * 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 GPIO53_SSP1SYSCLK_MD	(53 | 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 GPIO22_SSP2SYSCLK_MD	(22 | GPIO_ALT_FN_2_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 GPIO45_SSP3SYSCLK_MD	(45 | GPIO_ALT_FN_3_OUT)
++
++#if 0
++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),
++		.sys = GPIO53_SSP1SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO53_SSP1SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO53_SSP1SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO53_SSP1SYSCLK_MD,
++	}},
++	{{ /* 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),
++		.sys = GPIO22_SSP2SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO22_SSP2SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO22_SSP2SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO22_SSP2SYSCLK_MD,
++	}},
++	{{ /* 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),
++		.sys = GPIO45_SSP3SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO45_SSP3SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO45_SSP3SYSCLK_MD,
++	},
++	{ /* 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),
++		.sys = GPIO45_SSP3SYSCLK_MD,
++	}},
++};
++#endif
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_hifi_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int pll_out = 0, bclk = 0, fmt = 0;
++	int ret = 0;
++
++	/*
++	 * The WM8753 is far better at generating accurate audio clocks than the
++	 * pxa2xx I2S controller, so we will use it as master when we can.
++	 * i.e all rates except 8k and 16k as BCLK must be 64 * rate when the
++	 * pxa27x or pxa25x is slave. Note this restriction does not apply to SSP
++	 * I2S emulation mode.
++	 */
++	switch (params_rate(params)) {
++	case 8000:
++	case 16000:
++		fmt = SND_SOC_DAIFMT_CBS_CFS;
++		pll_out = 12288000;
++		break;
++	case 48000:
++		fmt = SND_SOC_DAIFMT_CBM_CFS;
++		bclk = WM8753_BCLK_DIV_4;
++		pll_out = 12288000;
++		break;
++	case 96000:
++		fmt = SND_SOC_DAIFMT_CBM_CFS;
++		bclk = WM8753_BCLK_DIV_2;
++		pll_out = 12288000;
++		break;
++	case 11025:
++		fmt = SND_SOC_DAIFMT_CBM_CFS;
++		bclk = WM8753_BCLK_DIV_16;
++		pll_out = 11289600;
++		break;
++	case 22050:
++		fmt = SND_SOC_DAIFMT_CBM_CFS;
++		bclk = WM8753_BCLK_DIV_8;
++		pll_out = 11289600;
++		break;
++	case 44100:
++		fmt = SND_SOC_DAIFMT_CBM_CFS;
++		bclk = WM8753_BCLK_DIV_4;
++		pll_out = 11289600;
++		break;
++	case 88200:
++		fmt = SND_SOC_DAIFMT_CBM_CFS;
++		bclk = WM8753_BCLK_DIV_2;
++		pll_out = 11289600;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai,
++		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
++		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set the I2S system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set codec BCLK division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
++	if (ret < 0)
++		return ret;
++
++	/* codec PLL input is 13 MHz */
++	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 13000000, pll_out);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static int mainstone_hifi_hw_free(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++
++	/* disable the PLL */
++	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
++}
++
++/*
++ * Mainstone WM8753 HiFi DAI opserations.
++ */
++static struct snd_soc_ops mainstone_hifi_ops = {
++	.hw_params = mainstone_hifi_hw_params,
++	.hw_free = mainstone_hifi_hw_free,
++};
++
++static int mainstone_voice_startup(struct snd_pcm_substream *substream)
++{
++	/* 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_voice_shutdown(struct snd_pcm_substream *substream)
++{
++//	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++	/* 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;
++
++	/* liam may need to tristate DAI */
++}
++
++static int mainstone_voice_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int pll_out = 0, bclk = 0, pcmdiv = 0;
++	int ret = 0;
++
++	/*
++	 * The WM8753 is far better at generating accurate audio clocks than the
++	 * pxa2xx SSP controller, so we will use it as master when we can.
++	 */
++	switch (params_rate(params)) {
++	case 8000:
++		pll_out = 12288000;
++		pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
++		bclk = WM8753_VXCLK_DIV_8; /* 256kHz */
++		break;
++	case 16000:
++		pll_out = 12288000;
++		pcmdiv = WM8753_PCM_DIV_3; /* 4.096 MHz */
++		bclk = WM8753_VXCLK_DIV_8; /* 512kHz */
++		break;
++	case 48000:
++		pll_out = 12288000;
++		pcmdiv = WM8753_PCM_DIV_1; /* 12.288 MHz */
++		bclk = WM8753_VXCLK_DIV_8; /* 1.536 MHz */
++		break;
++	case 11025:
++		pll_out = 11289600;
++		pcmdiv = WM8753_PCM_DIV_4; /* 11.2896 MHz */
++		bclk = WM8753_VXCLK_DIV_8; /* 352.8 kHz */
++		break;
++	case 22050:
++		pll_out = 11289600;
++		pcmdiv = WM8753_PCM_DIV_2; /* 11.2896 MHz */
++		bclk = WM8753_VXCLK_DIV_8; /* 705.6 kHz */
++		break;
++	case 44100:
++		pll_out = 11289600;
++		pcmdiv = WM8753_PCM_DIV_1; /* 11.2896 MHz */
++		bclk = WM8753_VXCLK_DIV_8; /* 1.4112 MHz */
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, pll_out,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set the SSP system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_PLL, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set codec BCLK division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_VXCLKDIV, bclk);
++	if (ret < 0)
++		return ret;
++
++	/* set codec PCM division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
++	if (ret < 0)
++		return ret;
++
++	/* codec PLL input is 13 MHz */
++	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 13000000, pll_out);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static int mainstone_voice_hw_free(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++
++	/* disable the PLL */
++	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
++}
++
++static struct snd_soc_ops mainstone_voice_ops = {
++	.startup = mainstone_voice_startup,
++	.shutdown = mainstone_voice_shutdown,
++	.hw_params = mainstone_voice_hw_params,
++	.hw_free = mainstone_voice_hw_free,
++};
++
++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;
++}
++
++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,
++	.ops = &mainstone_hifi_ops,
++},
++{ /* Voice via BT */
++	.name = "Bluetooth",
++	.stream_name = "Voice",
++	.cpu_dai = &pxa_ssp_dai[1],
++	.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
++	.ops = &mainstone_voice_ops,
++},
++};
++
++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 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);
++
++	/* SSP port 2 slave */
++	pxa_gpio_mode(GPIO11_SSP2RX_MD);
++	pxa_gpio_mode(GPIO13_SSP2TX_MD);
++	pxa_gpio_mode(GPIO22_SSP2CLKS_MD);
++	pxa_gpio_mode(GPIO88_SSP2FRMS_MD);
++
++	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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM8753 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8974.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8974.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,104 @@
++/*
++ * mainstone.c  --  SoC audio for Mainstone
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at 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"
++#include "pxa2xx-i2s.h"
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_wm8974_init(struct snd_soc_codec *codec)
++{
++	return 0;
++}
++
++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,
++},
++};
++
++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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9712.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9712.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,172 @@
++/*
++ * mainstone.c  --  SoC audio for Mainstone
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at 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"
++#include "pxa2xx-ac97.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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM9712 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9713.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9713.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,318 @@
++/*
++ * mainstone.c  --  SoC audio for Mainstone
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at 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"
++#include "pxa2xx-ac97.h"
++#include "pxa2xx-ssp.h"
++
++#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 GPIO22_SSP2SYSCLK_MD	(22 | GPIO_ALT_FN_2_OUT)
++
++static struct snd_soc_machine mainstone;
++
++static int mainstone_voice_startup(struct snd_pcm_substream *substream)
++{
++	/* 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_voice_shutdown(struct snd_pcm_substream *substream)
++{
++	/* 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 int mainstone_voice_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int bclk = 0, pcmdiv = 0;
++	int ret = 0;
++
++	switch (params_rate(params)) {
++	case 8000:
++		pcmdiv = WM9713_PCMDIV(12); /* 2.048 MHz */
++		bclk = WM9713_PCMBCLK_DIV_16; /* 128kHz */
++		break;
++	case 16000:
++		pcmdiv = WM9713_PCMDIV(6); /* 4.096 MHz */
++		bclk = WM9713_PCMBCLK_DIV_16; /* 256kHz */
++		break;
++	case 48000:
++		pcmdiv = WM9713_PCMDIV(2); /* 12.288 MHz */
++		bclk = WM9713_PCMBCLK_DIV_16; /* 512kHz */
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++	if (ret < 0)
++		return ret;
++
++	/* set the SSP system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_PLL, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set codec BCLK division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM9713_PCMBCLK_DIV, bclk);
++	if (ret < 0)
++		return ret;
++
++	/* set codec PCM division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, pcmdiv);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static struct snd_soc_ops mainstone_voice_ops = {
++	.startup = mainstone_voice_startup,
++	.shutdown = mainstone_voice_shutdown,
++	.hw_params = mainstone_voice_hw_params,
++};
++
++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;
++}
++
++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 = "WM9713",
++	.stream_name = "WM9713 Voice",
++	.cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2],
++	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
++	.ops = &mainstone_voice_ops,
++},
++};
++
++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_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);
++
++	/* SSP port 2 slave */
++	pxa_gpio_mode(GPIO11_SSP2RX_MD);
++	pxa_gpio_mode(GPIO13_SSP2TX_MD);
++	pxa_gpio_mode(GPIO22_SSP2CLKS_MD);
++	pxa_gpio_mode(GPIO88_SSP2FRMS_MD);
++
++	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 at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM9713 Mainstone");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/poodle.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/poodle.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,352 @@
++/*
++ * poodle.c  --  SoC audio for Poodle
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *          Richard Purdie <richard at 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"
++#include "pxa2xx-i2s.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 int poodle_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int clk = 0;
++	int ret = 0;
++
++	switch (params_rate(params)) {
++	case 8000:
++	case 16000:
++	case 48000:
++	case 96000:
++		clk = 12288000;
++		break;
++	case 11025:
++	case 22050:
++	case 44100:
++		clk = 11289600;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set the I2S system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static struct snd_soc_ops poodle_ops = {
++	.startup = poodle_startup,
++	.hw_params = poodle_hw_params,
++	.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;
++}
++
++/* 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,
++	.ops = &poodle_ops,
++};
++
++/* poodle audio machine driver */
++static struct snd_soc_machine snd_soc_machine_poodle = {
++	.name = "Poodle",
++	.dai_link = &poodle_dai,
++	.num_links = 1,
++};
++
++/* 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-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,431 @@
++/*
++ * 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"
++#include "pxa2xx-ac97.h"
++
++static DEFINE_MUTEX(car_mutex);
++static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
++static volatile long gsr_bits;
++
++/*
++ * 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;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
++	else
++		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;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
++	else
++		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;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		return -ENODEV;
++	else
++		cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
++
++	return 0;
++}
++
++#define PXA2XX_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)
++
++/*
++ * 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,
++		.rates = PXA2XX_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "AC97 Capture",
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = PXA2XX_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = pxa2xx_ac97_hw_params,},
++},
++{
++	.name = "pxa2xx-ac97-aux",
++	.id = 1,
++	.type = SND_SOC_DAI_AC97,
++	.playback = {
++		.stream_name = "AC97 Aux Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = PXA2XX_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "AC97 Aux Capture",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = PXA2XX_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = pxa2xx_ac97_hw_aux_params,},
++},
++{
++	.name = "pxa2xx-ac97-mic",
++	.id = 2,
++	.type = SND_SOC_DAI_AC97,
++	.capture = {
++		.stream_name = "AC97 Mic Capture",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = PXA2XX_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = pxa2xx_ac97_hw_mic_params,},
++},
++};
++
++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-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,318 @@
++/*
++ * pxa2xx-i2s.c  --  ALSA Soc Audio Layer
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  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"
++#include "pxa2xx-i2s.h"
++
++struct pxa_i2s_port {
++	u32 sadiv;
++	u32 sacr0;
++	u32 sacr1;
++	u32 saimr;
++	int master;
++	u32 fmt;
++};
++static struct pxa_i2s_port pxa_i2s;
++
++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;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (!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_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int fmt)
++{
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		pxa_i2s.fmt = 0;
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		pxa_i2s.fmt = SACR1_AMSL;
++		break;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBS_CFS:
++		pxa_i2s.master = 1;
++		break;
++	case SND_SOC_DAIFMT_CBM_CFS:
++		pxa_i2s.master = 0;
++		break;
++	default:
++		break;
++	}
++	return 0;
++}
++
++static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	if (clk_id != PXA2XX_I2S_SYSCLK)
++		return -ENODEV;
++
++	if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
++		pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
++
++	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;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	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)
++		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
++	else
++		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);
++		SACR1 |= pxa_i2s.fmt;
++	}
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		SAIMR |= SAIMR_TFS;
++	else
++		SAIMR |= SAIMR_RFS;
++
++	switch (params_rate(params)) {
++	case 8000:
++		SADIV = 0x48;
++		break;
++	case 11025:
++		SADIV = 0x34;
++		break;
++	case 16000:
++		SADIV = 0x24;
++		break;
++	case 22050:
++		SADIV = 0x1a;
++		break;
++	case 44100:
++		SADIV = 0xd;
++		break;
++	case 48000:
++		SADIV = 0xc;
++		break;
++	case 96000: /* not in manual and possibly slightly inaccurate */
++		SADIV = 0x6;
++		break;
++	}
++
++	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
++
++#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
++
++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,
++	.playback = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = PXA2XX_I2S_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = PXA2XX_I2S_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.startup = pxa2xx_i2s_startup,
++		.shutdown = pxa2xx_i2s_shutdown,
++		.trigger = pxa2xx_i2s_trigger,
++		.hw_params = pxa2xx_i2s_hw_params,},
++	.dai_ops = {
++		.set_fmt = pxa2xx_i2s_set_dai_fmt,
++		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
++	},
++};
++
++EXPORT_SYMBOL_GPL(pxa_i2s_dai);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,368 @@
++/*
++ * 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->dai->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;
++
++	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
++	if (ret < 0)
++		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;
++//printk("%s %d prtd %p\n", __FUNCTION__, __LINE__, prtd);
++	dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
++			      prtd->dma_desc_array, prtd->dma_desc_array_phys);
++	kfree(prtd);
++//	runtime->private_data = NULL;// liam ????
++	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-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,34 @@
++/*
++ * 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;
++};
++
++/* platform data */
++extern struct snd_soc_platform pxa2xx_soc_platform;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,671 @@
++/*
++ * pxa2xx-ssp.c  --  ALSA Soc Audio Layer
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    12th Aug 2005   Initial version.
++ *
++ * TODO:
++ *  o The SSP driver _mostly_ works, however is in need of testing and
++ *     someone with time to complete it.
++ *  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/pcm_params.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"
++#include "pxa2xx-ssp.h"
++
++#define PXA_SSP_DEBUG 1
++
++/*
++ * The following should be defined in pxa-regs.h
++ */
++#define SSCR0_ACS		(1 << 30)	/* Audio Clock Select */
++#define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass (SSCR0[ACS] must be set) */
++#define SSACD_ACPS(x)	(x << 4)	/* Audio clock PLL select */
++#define SSACD_ACDS(x)	(x << 0)	/* Audio clock divider select */
++
++/*
++ * SSP audio private data
++ */
++struct ssp_priv {
++	unsigned int sysclk;
++};
++
++static struct ssp_priv ssp_clk[3];
++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 int pxa2xx_ssp_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	int ret = 0;
++
++	if (!rtd->dai->cpu_dai->active) {
++		ret = ssp_init (&ssp[cpu_dai->id], cpu_dai->id + 1,
++			SSP_NO_IRQ);
++		if (ret < 0)
++			return ret;
++		ssp_disable(&ssp[cpu_dai->id]);
++	}
++	return ret;
++}
++
++static void pxa2xx_ssp_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (!cpu_dai->active) {
++		ssp_disable(&ssp[cpu_dai->id]);
++		ssp_exit(&ssp[cpu_dai->id]);
++	}
++}
++
++#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
++
++#ifdef CONFIG_PM
++
++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
++
++/*
++ * Set the SSP ports SYSCLK.
++ */
++static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
++	int clk_id, unsigned int freq, int dir)
++{
++	int port = cpu_dai->id + 1;
++	u32 sscr0 = SSCR0_P(port) &
++		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
++
++	switch (clk_id) {
++	case PXA2XX_SSP_CLK_PLL:
++		/* Internal PLL is fixed on pxa25x and pxa27x */
++#ifdef CONFIG_PXA27x
++		ssp_clk[cpu_dai->id].sysclk = 13000000;
++#else
++		ssp_clk[cpu_dai->id].sysclk = 1843200;
++#endif
++		break;
++	case PXA2XX_SSP_CLK_EXT:
++		ssp_clk[cpu_dai->id].sysclk = freq;
++		sscr0 |= SSCR0_ECS;
++		break;
++	case PXA2XX_SSP_CLK_NET:
++		ssp_clk[cpu_dai->id].sysclk = freq;
++		sscr0 |= SSCR0_NCS | SSCR0_MOD;
++		break;
++	case PXA2XX_SSP_CLK_AUDIO:
++		ssp_clk[cpu_dai->id].sysclk = 0;
++		SSCR0_P(port) |= SSCR0_SerClkDiv(1);
++		sscr0 |= SSCR0_ACS;
++		break;
++	default:
++		return -ENODEV;
++	}
++
++	/* the SSP CKEN clock must be disabled when changing SSP clock mode */
++	pxa_set_cken(cken[cpu_dai->id], 0);
++	SSCR0_P(port) |= sscr0;
++	pxa_set_cken(cken[cpu_dai->id], 1);
++	return 0;
++}
++
++/*
++ * Set the SSP clock dividers.
++ */
++static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
++	int div_id, int div)
++{
++	int port = cpu_dai->id + 1;
++
++	switch (div_id) {
++	case PXA2XX_SSP_AUDIO_DIV_ACDS:
++		SSACD_P(port) &= ~ 0x7;
++		SSACD_P(port) |= SSACD_ACDS(div);
++		break;
++	case PXA2XX_SSP_AUDIO_DIV_SCDB:
++		SSACD_P(port) &= ~0x8;
++		if (div == PXA2XX_SSP_CLK_SCDB_1)
++			SSACD_P(port) |= SSACD_SCDB;
++		break;
++	case PXA2XX_SSP_DIV_SCR:
++		SSCR0_P(port) &= ~SSCR0_SCR;
++		SSCR0_P(port) |= SSCR0_SerClkDiv(div);
++		break;
++	default:
++		return -ENODEV;
++	}
++
++	return 0;
++}
++
++/*
++ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
++ */
++static int pxa2xx_ssp_set_dai_pll(struct snd_soc_cpu_dai *cpu_dai,
++	int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	int port = cpu_dai->id + 1;
++
++	SSACD_P(port) &= ~0x70;
++	switch (freq_out) {
++	case 5622000:
++		break;
++	case 11345000:
++		SSACD_P(port) |= (0x1 << 4);
++		break;
++	case 12235000:
++		SSACD_P(port) |= (0x2 << 4);
++		break;
++	case 14857000:
++		SSACD_P(port) |= (0x3 << 4);
++		break;
++	case 32842000:
++		SSACD_P(port) |= (0x4 << 4);
++		break;
++	case 48000000:
++		SSACD_P(port) |= (0x5 << 4);
++		break;
++	}
++	return 0;
++}
++
++/*
++ * Set the active slots in TDM/Network mode
++ */
++static int pxa2xx_ssp_set_dai_tdm_slot(struct snd_soc_cpu_dai *cpu_dai,
++	unsigned int mask, int slots)
++{
++	int port = cpu_dai->id + 1;
++
++	SSCR0_P(port) &= ~SSCR0_SlotsPerFrm(7);
++
++	/* set number of active slots */
++	SSCR0_P(port) |= SSCR0_SlotsPerFrm(slots);
++
++	/* set active slot mask */
++	SSTSA_P(port) = mask;
++	SSRSA_P(port) = mask;
++	return 0;
++}
++
++/*
++ * Tristate the SSP DAI lines
++ */
++static int pxa2xx_ssp_set_dai_tristate(struct snd_soc_cpu_dai *cpu_dai,
++	int tristate)
++{
++	int port = cpu_dai->id + 1;
++
++	if (tristate)
++		SSCR1_P(port) &= ~SSCR1_TTE;
++	else
++		SSCR1_P(port) |= SSCR1_TTE;
++
++	return 0;
++}
++
++/*
++ * Set up the SSP DAI format.
++ * The SSP Port must be inactive before calling this function as the
++ * physical interface format is changed.
++ */
++static int pxa2xx_ssp_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int fmt)
++{
++	int port = cpu_dai->id + 1;
++
++	/* reset port settings */
++	SSCR0_P(port) = 0;
++	SSCR1_P(port) = 0;
++	SSPSP_P(port) = 0;
++
++	/* NOTE: I2S emulation is still very much work in progress here */
++
++	/* FIXME: this is what wince uses for msb */
++	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_MSB) {
++		SSCR0_P(port) = SSCR0_EDSS | SSCR0_TISSP | SSCR0_DataSize(16);
++
++//		SSCR1_P(port) = SSCR1_RxTresh(8) | SSCR1_TxTresh(8); /* doesn't seem to be needed */
++		return 0;
++	}
++
++	/* check for I2S emulation mode - handle it separately  */
++	if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) ||
++		((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_MSB)) {
++		/* 8.4.11 */
++
++		/* Only SSCR0[NCS] or SSCR0[ECS] bit fields settings are optional */
++		SSCR0_P(port) = SSCR0_EDSS | SSCR0_PSP | SSCR0_DataSize(16);
++
++		/* SSCR1 = 0x203C3C03 */
++		/* SSCR1[SCLKDIR] and SSCR1[SFRMDIR] must be cleared (master only ???),
++		 * all other bit fields settings are optional. */
++		//SSCR1_P(port) &= ~(SSCR1_SCLKDIR | SSCR1_SFRMDIR);
++
++		/* set FIFO thresholds */
++		SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1);
++
++		/* normal: */
++		/* all bit fields must be cleared except: FSRT = 1 and
++		 * SFRMWDTH = 16, DMYSTART=0,1) */
++		SSPSP_P(port) = SSPSP_FSRT | SSPSP_SFRMWDTH(16) | SSPSP_DMYSTRT(0);
++		return 0;
++	}
++
++	SSCR0_P(port) |= SSCR0_PSP;
++	SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) |
++		SSCR1_TRAIL | SSCR1_RWOT;
++
++	switch(fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		SSCR1_P(port) |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR);
++		break;
++	case SND_SOC_DAIFMT_CBM_CFS:
++		SSCR1_P(port) |= SSCR1_SCLKDIR;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFM:
++		SSCR1_P(port) |= SSCR1_SFRMDIR;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
++	case SND_SOC_DAIFMT_NB_NF:
++		SSPSP_P(port) |= SSPSP_SFRMP | SSPSP_FSRT;
++		break;
++	case SND_SOC_DAIFMT_IB_IF:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_DSP_A:
++		SSPSP_P(port) |= SSPSP_DMYSTRT(1);
++	case SND_SOC_DAIFMT_DSP_B:
++		SSPSP_P(port) |= SSPSP_SCMODE(2);
++		break;
++	case SND_SOC_DAIFMT_I2S:
++	case SND_SOC_DAIFMT_MSB:
++		/* handled above */
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++/*
++ * Set the SSP audio DMA parameters and sample size.
++ * Can be called multiple times by oss emulation.
++ */
++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;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	int dma = 0, chn = params_channels(params);
++	int port = cpu_dai->id + 1;
++
++	/* select correct DMA params */
++	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
++		dma = 1; /* capture DMA offset is 1,3 */
++	if (chn == 2)
++		dma += 2; /* stereo DMA offset is 2, mono is 0 */
++	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
++
++	/* we can only change the settings if the port is not in use */
++	if (SSCR0_P(port) & SSCR0_SSE)
++		return 0;
++
++	/* clear selected SSP bits */
++	SSCR0_P(port) &= ~(SSCR0_DSS | SSCR0_EDSS);
++
++	/* bit size */
++	switch(params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		SSCR0_P(port) |= SSCR0_DataSize(16);
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		SSCR0_P(port) |=(SSCR0_EDSS | SSCR0_DataSize(8));
++		/* we must be in network mode (2 slots) for 24 bit stereo */
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		SSCR0_P(port) |= (SSCR0_EDSS | SSCR0_DataSize(16));
++		/* we must be in network mode (2 slots) for 32 bit stereo */
++		break;
++	}
++
++#if PXA_SSP_DEBUG
++	printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x SSACD %x\n",
++		SSCR0_P(port), SSCR1_P(port),
++		SSTO_P(port), SSPSP_P(port),
++		SSSR_P(port), SSACD_P(port));
++#endif
++	return 0;
++}
++
++static int pxa2xx_ssp_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	int ret = 0;
++	int port = cpu_dai->id + 1;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_RESUME:
++		ssp_enable(&ssp[cpu_dai->id]);
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			SSCR1_P(port) |= SSCR1_TSRE;
++		else
++			SSCR1_P(port) |= SSCR1_RSRE;
++		SSSR_P(port) |= SSSR_P(port);
++		break;
++	case SNDRV_PCM_TRIGGER_START:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			SSCR1_P(port) |= SSCR1_TSRE;
++		else
++			SSCR1_P(port) |= SSCR1_RSRE;
++		ssp_enable(&ssp[cpu_dai->id]);
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			SSCR1_P(port) &= ~SSCR1_TSRE;
++		else
++			SSCR1_P(port) &= ~SSCR1_RSRE;
++		break;
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++		ssp_disable(&ssp[cpu_dai->id]);
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			SSCR1_P(port) &= ~SSCR1_TSRE;
++		else
++			SSCR1_P(port) &= ~SSCR1_RSRE;
++		break;
++
++	default:
++		ret = -EINVAL;
++	}
++#if PXA_SSP_DEBUG
++	printk("trig cmd %d\n", cmd);
++	printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n",
++		SSCR0_P(port), SSCR1_P(port),
++		SSTO_P(port), SSPSP_P(port),
++		SSSR_P(port));
++#endif
++	return ret;
++}
++
++#define PXA2XX_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
++	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++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,
++		.playback = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = PXA2XX_SSP_RATES,
++			.formats = PXA2XX_SSP_FORMATS,},
++		.capture = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = PXA2XX_SSP_RATES,
++			.formats = PXA2XX_SSP_FORMATS,},
++		.ops = {
++			.startup = pxa2xx_ssp_startup,
++			.shutdown = pxa2xx_ssp_shutdown,
++			.trigger = pxa2xx_ssp_trigger,
++			.hw_params = pxa2xx_ssp_hw_params,},
++		.dai_ops = {
++			.set_sysclk = pxa2xx_ssp_set_dai_sysclk,
++			.set_clkdiv = pxa2xx_ssp_set_dai_clkdiv,
++			.set_pll = pxa2xx_ssp_set_dai_pll,
++			.set_fmt = pxa2xx_ssp_set_dai_fmt,
++			.set_tdm_slot = pxa2xx_ssp_set_dai_tdm_slot,
++			.set_tristate = pxa2xx_ssp_set_dai_tristate,
++		},
++	},
++	{	.name = "pxa2xx-ssp2",
++		.id = 1,
++		.type = SND_SOC_DAI_PCM,
++		.suspend = pxa2xx_ssp_suspend,
++		.resume = pxa2xx_ssp_resume,
++		.playback = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = PXA2XX_SSP_RATES,
++			.formats = PXA2XX_SSP_FORMATS,},
++		.capture = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = PXA2XX_SSP_RATES,
++			.formats = PXA2XX_SSP_FORMATS,},
++		.ops = {
++			.startup = pxa2xx_ssp_startup,
++			.shutdown = pxa2xx_ssp_shutdown,
++			.trigger = pxa2xx_ssp_trigger,
++			.hw_params = pxa2xx_ssp_hw_params,},
++		.dai_ops = {
++			.set_sysclk = pxa2xx_ssp_set_dai_sysclk,
++			.set_clkdiv = pxa2xx_ssp_set_dai_clkdiv,
++			.set_pll = pxa2xx_ssp_set_dai_pll,
++			.set_fmt = pxa2xx_ssp_set_dai_fmt,
++			.set_tdm_slot = pxa2xx_ssp_set_dai_tdm_slot,
++			.set_tristate = pxa2xx_ssp_set_dai_tristate,
++		},
++	},
++	{	.name = "pxa2xx-ssp3",
++		.id = 2,
++		.type = SND_SOC_DAI_PCM,
++		.suspend = pxa2xx_ssp_suspend,
++		.resume = pxa2xx_ssp_resume,
++		.playback = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = PXA2XX_SSP_RATES,
++			.formats = PXA2XX_SSP_FORMATS,},
++		.capture = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = PXA2XX_SSP_RATES,
++			.formats = PXA2XX_SSP_FORMATS,},
++		.ops = {
++			.startup = pxa2xx_ssp_startup,
++			.shutdown = pxa2xx_ssp_shutdown,
++			.trigger = pxa2xx_ssp_trigger,
++			.hw_params = pxa2xx_ssp_hw_params,},
++		.dai_ops = {
++			.set_sysclk = pxa2xx_ssp_set_dai_sysclk,
++			.set_clkdiv = pxa2xx_ssp_set_dai_clkdiv,
++			.set_pll = pxa2xx_ssp_set_dai_pll,
++			.set_fmt = pxa2xx_ssp_set_dai_fmt,
++			.set_tdm_slot = pxa2xx_ssp_set_dai_tdm_slot,
++			.set_tristate = pxa2xx_ssp_set_dai_tristate,
++		},
++	},
++};
++EXPORT_SYMBOL_GPL(pxa_ssp_dai);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("pxa2xx SSP/PCM SoC Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/spitz.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/spitz.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,394 @@
++/*
++ * 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 at wolfsonmicro.com>
++ *          Richard Purdie <richard at 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"
++#include "pxa2xx-i2s.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 int spitz_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int clk = 0;
++	int ret = 0;
++
++	switch (params_rate(params)) {
++	case 8000:
++	case 16000:
++	case 48000:
++	case 96000:
++		clk = 12288000;
++		break;
++	case 11025:
++	case 22050:
++	case 44100:
++		clk = 11289600;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set the I2S system clock as input (unused) */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static struct snd_soc_ops spitz_ops = {
++	.startup = spitz_startup,
++	.hw_params = spitz_hw_params,
++};
++
++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;
++}
++
++/* 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,
++	.ops = &spitz_ops,
++};
++
++/* spitz audio machine driver */
++static struct snd_soc_machine snd_soc_machine_spitz = {
++	.name = "Spitz",
++	.dai_link = &spitz_dai,
++	.num_links = 1,
++};
++
++/* 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-2.6.17.14-fic4.test/sound/soc/pxa/tosa.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/tosa.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,289 @@
++/*
++ * tosa.c  --  SoC audio for Tosa
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *          Richard Purdie <richard at 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"
++#include "pxa2xx-ac97.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),
++};
++
++static const struct snd_kcontrol_new tosa_controls[] = {
++	SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
++		tosa_set_jack),
++	SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
++		tosa_set_spk),
++};
++
++static int tosa_ac97_init(struct snd_soc_codec *codec)
++{
++	int i, err;
++
++	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
++	snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
++
++	/* add tosa specific controls */
++	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&tosa_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	/* add tosa specific widgets */
++	for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
++	}
++
++	/* set up tosa 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 struct snd_soc_dai_link tosa_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 = tosa_ac97_init,
++	.ops = &tosa_ops,
++},
++{
++	.name = "AC97 Aux",
++	.stream_name = "AC97 Aux",
++	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
++	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
++	.ops = &tosa_ops,
++},
++};
++
++static struct snd_soc_machine tosa = {
++	.name = "Tosa",
++	.dai_link = tosa_dai,
++	.num_links = ARRAY_SIZE(tosa_dai),
++};
++
++static struct snd_soc_device tosa_snd_devdata = {
++	.machine = &tosa,
++	.platform = &pxa2xx_soc_platform,
++	.codec_dev = &soc_codec_dev_wm9712,
++};
++
++static struct platform_device *tosa_snd_device;
++
++static int __init tosa_init(void)
++{
++	int ret;
++
++	if (!machine_is_tosa())
++		return -ENODEV;
++
++	tosa_snd_device = platform_device_alloc("soc-audio", -1);
++	if (!tosa_snd_device)
++		return -ENOMEM;
++
++	platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata);
++	tosa_snd_devdata.dev = &tosa_snd_device->dev;
++	ret = platform_device_add(tosa_snd_device);
++
++	if (ret)
++		platform_device_put(tosa_snd_device);
++
++	return ret;
++}
++
++static void __exit tosa_exit(void)
++{
++	platform_device_unregister(tosa_snd_device);
++}
++
++module_init(tosa_init);
++module_exit(tosa_exit);
++
++/* Module information */
++MODULE_AUTHOR("Richard Purdie");
++MODULE_DESCRIPTION("ALSA SoC Tosa");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/soc-dapm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/soc-dapm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,1329 @@
++/*
++ * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    12th Aug 2005   Initial version.
++ *    25th Oct 2005   Implemented path power domain.
++ *    18th Dec 2005   Implemented machine and stream level power domain.
++ *
++ *  Features:
++ *    o Changes power status of internal codec blocks depending on the
++ *      dynamic configuration of codec internal audio paths and active
++ *      DAC's/ADC's.
++ *    o Platform power domain - can support external components i.e. amps and
++ *      mic/meadphone insertion events.
++ *    o Automatic Mic Bias support
++ *    o Jack insertion power event initiation - e.g. hp insertion will enable
++ *      sinks, dacs, etc
++ *    o Delayed powerdown of audio susbsytem to reduce pops between a quick
++ *      device reopen.
++ *
++ *  Todo:
++ *    o DAPM power change sequencing - allow for configurable per
++ *      codec sequences.
++ *    o Support for analogue bias optimisation.
++ *    o Support for reduced codec oversampling rates.
++ *    o Support for reduced codec bias currents.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/bitops.h>
++#include <linux/platform_device.h>
++#include <linux/jiffies.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++/* debug */
++#define DAPM_DEBUG 0
++#if DAPM_DEBUG
++#define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
++#define dbg(format, arg...) printk(format, ## arg)
++#else
++#define dump_dapm(codec, action)
++#define dbg(format, arg...)
++#endif
++
++#define POP_DEBUG 0
++#if POP_DEBUG
++#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */
++#define pop_wait(time) schedule_timeout_interruptible(msecs_to_jiffies(time))
++#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME)
++#else
++#define pop_dbg(format, arg...)
++#define pop_wait(time)
++#endif
++
++/* dapm power sequences - make this per codec in the future */
++static int dapm_up_seq[] = {
++	snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
++	snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga,
++	snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
++};
++static int dapm_down_seq[] = {
++	snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
++	snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
++	snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post
++};
++
++static int dapm_status = 1;
++module_param(dapm_status, int, 0);
++MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
++
++/* create a new dapm widget */
++static struct snd_soc_dapm_widget *dapm_cnew_widget(
++	const struct snd_soc_dapm_widget *_widget)
++{
++	struct snd_soc_dapm_widget* widget;
++	widget = kmalloc(sizeof(struct snd_soc_dapm_widget), GFP_KERNEL);
++	if (!widget)
++		return NULL;
++
++	memcpy(widget, _widget, sizeof(struct snd_soc_dapm_widget));
++	return widget;
++}
++
++/* set up initial codec paths */
++static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
++	struct snd_soc_dapm_path *p, int i)
++{
++	switch (w->id) {
++	case snd_soc_dapm_switch:
++	case snd_soc_dapm_mixer: {
++		int val;
++		int reg = w->kcontrols[i].private_value & 0xff;
++		int shift = (w->kcontrols[i].private_value >> 8) & 0x0f;
++		int mask = (w->kcontrols[i].private_value >> 16) & 0xff;
++		int invert = (w->kcontrols[i].private_value >> 24) & 0x01;
++
++		val = snd_soc_read(w->codec, reg);
++		val = (val >> shift) & mask;
++
++		if ((invert && !val) || (!invert && val))
++			p->connect = 1;
++		else
++			p->connect = 0;
++	}
++	break;
++	case snd_soc_dapm_mux: {
++		struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value;
++		int val, item, bitmask;
++
++		for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
++		;
++		val = snd_soc_read(w->codec, e->reg);
++		item = (val >> e->shift_l) & (bitmask - 1);
++
++		p->connect = 0;
++		for (i = 0; i < e->mask; i++) {
++			if (!(strcmp(p->name, e->texts[i])) && item == i)
++				p->connect = 1;
++		}
++	}
++	break;
++	/* does not effect routing - always connected */
++	case snd_soc_dapm_pga:
++	case snd_soc_dapm_output:
++	case snd_soc_dapm_adc:
++	case snd_soc_dapm_input:
++	case snd_soc_dapm_dac:
++	case snd_soc_dapm_micbias:
++	case snd_soc_dapm_vmid:
++		p->connect = 1;
++	break;
++	/* does effect routing - dynamically connected */
++	case snd_soc_dapm_hp:
++	case snd_soc_dapm_mic:
++	case snd_soc_dapm_spk:
++	case snd_soc_dapm_line:
++	case snd_soc_dapm_pre:
++	case snd_soc_dapm_post:
++		p->connect = 0;
++	break;
++	}
++}
++
++/* connect mux widget to it's interconnecting audio paths */
++static int dapm_connect_mux(struct snd_soc_codec *codec,
++	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
++	struct snd_soc_dapm_path *path, const char *control_name,
++	const struct snd_kcontrol_new *kcontrol)
++{
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++	int i;
++
++	for (i = 0; i < e->mask; i++) {
++		if (!(strcmp(control_name, e->texts[i]))) {
++			list_add(&path->list, &codec->dapm_paths);
++			list_add(&path->list_sink, &dest->sources);
++			list_add(&path->list_source, &src->sinks);
++			path->name = (char*)e->texts[i];
++			dapm_set_path_status(dest, path, 0);
++			return 0;
++		}
++	}
++
++	return -ENODEV;
++}
++
++/* connect mixer widget to it's interconnecting audio paths */
++static int dapm_connect_mixer(struct snd_soc_codec *codec,
++	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
++	struct snd_soc_dapm_path *path, const char *control_name)
++{
++	int i;
++
++	/* search for mixer kcontrol */
++	for (i = 0; i < dest->num_kcontrols; i++) {
++		if (!strcmp(control_name, dest->kcontrols[i].name)) {
++			list_add(&path->list, &codec->dapm_paths);
++			list_add(&path->list_sink, &dest->sources);
++			list_add(&path->list_source, &src->sinks);
++			path->name = dest->kcontrols[i].name;
++			dapm_set_path_status(dest, path, i);
++			return 0;
++		}
++	}
++	return -ENODEV;
++}
++
++/* update dapm codec register bits */
++static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
++{
++	int change, power;
++	unsigned short old, new;
++	struct snd_soc_codec *codec = widget->codec;
++
++	/* check for valid widgets */
++	if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
++		widget->id == snd_soc_dapm_output ||
++		widget->id == snd_soc_dapm_hp ||
++		widget->id == snd_soc_dapm_mic ||
++		widget->id == snd_soc_dapm_line ||
++		widget->id == snd_soc_dapm_spk)
++		return 0;
++
++	power = widget->power;
++	if (widget->invert)
++		power = (power ? 0:1);
++
++	old = snd_soc_read(codec, widget->reg);
++	new = (old & ~(0x1 << widget->shift)) | (power << widget->shift);
++
++	change = old != new;
++	if (change) {
++		pop_dbg("pop test %s : %s in %d ms\n", widget->name,
++			widget->power ? "on" : "off", POP_TIME);
++		snd_soc_write(codec, widget->reg, new);
++		pop_wait(POP_TIME);
++	}
++	dbg("reg old %x new %x change %d\n", old, new, change);
++	return change;
++}
++
++/* ramps the volume up or down to minimise pops before or after a
++ * DAPM power event */
++static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power)
++{
++	const struct snd_kcontrol_new *k = widget->kcontrols;
++
++	if (widget->muted && !power)
++		return 0;
++	if (!widget->muted && power)
++		return 0;
++
++	if (widget->num_kcontrols && k) {
++		int reg = k->private_value & 0xff;
++		int shift = (k->private_value >> 8) & 0x0f;
++		int mask = (k->private_value >> 16) & 0xff;
++		int invert = (k->private_value >> 24) & 0x01;
++
++		if (power) {
++			int i;
++			/* power up has happended, increase volume to last level */
++			if (invert) {
++				for (i = mask; i > widget->saved_value; i--)
++					snd_soc_update_bits(widget->codec, reg, mask, i);
++			} else {
++				for (i = 0; i < widget->saved_value; i++)
++					snd_soc_update_bits(widget->codec, reg, mask, i);
++			}
++			widget->muted = 0;
++		} else {
++			/* power down is about to occur, decrease volume to mute */
++			int val = snd_soc_read(widget->codec, reg);
++			int i = widget->saved_value = (val >> shift) & mask;
++			if (invert) {
++				for (; i < mask; i++)
++					snd_soc_update_bits(widget->codec, reg, mask, i);
++			} else {
++				for (; i > 0; i--)
++					snd_soc_update_bits(widget->codec, reg, mask, i);
++			}
++			widget->muted = 1;
++		}
++	}
++	return 0;
++}
++
++/* create new dapm mixer control */
++static int dapm_new_mixer(struct snd_soc_codec *codec,
++	struct snd_soc_dapm_widget *w)
++{
++	int i, ret = 0;
++	char name[32];
++	struct snd_soc_dapm_path *path;
++
++	/* add kcontrol */
++	for (i = 0; i < w->num_kcontrols; i++) {
++
++		/* match name */
++		list_for_each_entry(path, &w->sources, list_sink) {
++
++			/* mixer/mux paths name must match control name */
++			if (path->name != (char*)w->kcontrols[i].name)
++				continue;
++
++			/* add dapm control with long name */
++			snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
++			path->long_name = kstrdup (name, GFP_KERNEL);
++			if (path->long_name == NULL)
++				return -ENOMEM;
++
++			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
++				path->long_name);
++			ret = snd_ctl_add(codec->card, path->kcontrol);
++			if (ret < 0) {
++				printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
++						path->long_name);
++				kfree(path->long_name);
++				path->long_name = NULL;
++				return ret;
++			}
++		}
++	}
++	return ret;
++}
++
++/* create new dapm mux control */
++static int dapm_new_mux(struct snd_soc_codec *codec,
++	struct snd_soc_dapm_widget *w)
++{
++	struct snd_soc_dapm_path *path = NULL;
++	struct snd_kcontrol *kcontrol;
++	int ret = 0;
++
++	if (!w->num_kcontrols) {
++		printk(KERN_ERR "asoc: mux %s has no controls\n", w->name);
++		return -EINVAL;
++	}
++
++	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
++	ret = snd_ctl_add(codec->card, kcontrol);
++	if (ret < 0)
++		goto err;
++
++	list_for_each_entry(path, &w->sources, list_sink)
++		path->kcontrol = kcontrol;
++
++	return ret;
++
++err:
++	printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
++	return ret;
++}
++
++/* create new dapm volume control */
++static int dapm_new_pga(struct snd_soc_codec *codec,
++	struct snd_soc_dapm_widget *w)
++{
++	struct snd_kcontrol *kcontrol;
++	int ret = 0;
++
++	if (!w->num_kcontrols)
++		return -EINVAL;
++
++	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
++	ret = snd_ctl_add(codec->card, kcontrol);
++	if (ret < 0) {
++		printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
++		return ret;
++	}
++
++	return ret;
++}
++
++/* reset 'walked' bit for each dapm path */
++static inline void dapm_clear_walk(struct snd_soc_codec *codec)
++{
++	struct snd_soc_dapm_path *p;
++
++	list_for_each_entry(p, &codec->dapm_paths, list)
++		p->walked = 0;
++}
++
++/*
++ * Recursively check for a completed path to an active or physically connected
++ * output widget. Returns number of complete paths.
++ */
++static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
++{
++	struct snd_soc_dapm_path *path;
++	int con = 0;
++
++	if (widget->id == snd_soc_dapm_adc && widget->active)
++		return 1;
++
++	if (widget->connected) {
++		/* connected pin ? */
++		if (widget->id == snd_soc_dapm_output && !widget->ext)
++			return 1;
++
++		/* connected jack or spk ? */
++		if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
++			widget->id == snd_soc_dapm_line)
++			return 1;
++	}
++
++	list_for_each_entry(path, &widget->sinks, list_source) {
++		if (path->walked)
++			continue;
++
++		if (path->sink && path->connect) {
++			path->walked = 1;
++			con += is_connected_output_ep(path->sink);
++		}
++	}
++
++	return con;
++}
++
++/*
++ * Recursively check for a completed path to an active or physically connected
++ * input widget. Returns number of complete paths.
++ */
++static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
++{
++	struct snd_soc_dapm_path *path;
++	int con = 0;
++
++	/* active stream ? */
++	if (widget->id == snd_soc_dapm_dac && widget->active)
++		return 1;
++
++	if (widget->connected) {
++		/* connected pin ? */
++		if (widget->id == snd_soc_dapm_input && !widget->ext)
++			return 1;
++
++		/* connected VMID/Bias for lower pops */
++		if (widget->id == snd_soc_dapm_vmid)
++			return 1;
++
++		/* connected jack ? */
++		if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
++			return 1;
++	}
++
++	list_for_each_entry(path, &widget->sources, list_sink) {
++		if (path->walked)
++			continue;
++
++		if (path->source && path->connect) {
++			path->walked = 1;
++			con += is_connected_input_ep(path->source);
++		}
++	}
++
++	return con;
++}
++
++/*
++ * Scan each dapm widget for complete audio path.
++ * A complete path is a route that has valid endpoints i.e.:-
++ *
++ *  o DAC to output pin.
++ *  o Input Pin to ADC.
++ *  o Input pin to Output pin (bypass, sidetone)
++ *  o DAC to ADC (loopback).
++ */
++static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
++{
++	struct snd_soc_dapm_widget *w;
++	int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
++
++	/* do we have a sequenced stream event */
++	if (event == SND_SOC_DAPM_STREAM_START) {
++		c = ARRAY_SIZE(dapm_up_seq);
++		seq = dapm_up_seq;
++	} else if (event == SND_SOC_DAPM_STREAM_STOP) {
++		c = ARRAY_SIZE(dapm_down_seq);
++		seq = dapm_down_seq;
++	}
++
++	for(i = 0; i < c; i++) {
++		list_for_each_entry(w, &codec->dapm_widgets, list) {
++
++			/* is widget in stream order */
++			if (seq && seq[i] && w->id != seq[i])
++				continue;
++
++			/* vmid - no action */
++			if (w->id == snd_soc_dapm_vmid)
++				continue;
++
++			/* active ADC */
++			if (w->id == snd_soc_dapm_adc && w->active) {
++				in = is_connected_input_ep(w);
++				dapm_clear_walk(w->codec);
++				w->power = (in != 0) ? 1 : 0;
++				dapm_update_bits(w);
++				continue;
++			}
++
++			/* active DAC */
++			if (w->id == snd_soc_dapm_dac && w->active) {
++				out = is_connected_output_ep(w);
++				dapm_clear_walk(w->codec);
++				w->power = (out != 0) ? 1 : 0;
++				dapm_update_bits(w);
++				continue;
++			}
++
++			/* programmable gain/attenuation */
++			if (w->id == snd_soc_dapm_pga) {
++				int on;
++				in = is_connected_input_ep(w);
++				dapm_clear_walk(w->codec);
++				out = is_connected_output_ep(w);
++				dapm_clear_walk(w->codec);
++				w->power = on = (out != 0 && in != 0) ? 1 : 0;
++
++				if (!on)
++					dapm_set_pga(w, on); /* lower volume to reduce pops */
++				dapm_update_bits(w);
++				if (on)
++					dapm_set_pga(w, on); /* restore volume from zero */
++
++				continue;
++			}
++
++			/* pre and post event widgets */
++			if (w->id == snd_soc_dapm_pre) {
++				if (!w->event)
++					continue;
++
++				if (event == SND_SOC_DAPM_STREAM_START) {
++					ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
++					if (ret < 0)
++						return ret;
++				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
++					ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
++					if (ret < 0)
++						return ret;
++				}
++				continue;
++			}
++			if (w->id == snd_soc_dapm_post) {
++				if (!w->event)
++					continue;
++
++				if (event == SND_SOC_DAPM_STREAM_START) {
++					ret = w->event(w, SND_SOC_DAPM_POST_PMU);
++					if (ret < 0)
++						return ret;
++				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
++					ret = w->event(w, SND_SOC_DAPM_POST_PMD);
++					if (ret < 0)
++						return ret;
++				}
++				continue;
++			}
++
++			/* all other widgets */
++			in = is_connected_input_ep(w);
++			dapm_clear_walk(w->codec);
++			out = is_connected_output_ep(w);
++			dapm_clear_walk(w->codec);
++			power = (out != 0 && in != 0) ? 1 : 0;
++			power_change = (w->power == power) ? 0: 1;
++			w->power = power;
++
++			/* call any power change event handlers */
++			if (power_change) {
++				if (w->event) {
++					dbg("power %s event for %s flags %x\n",
++						w->power ? "on" : "off", w->name, w->event_flags);
++					if (power) {
++						/* power up event */
++						if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
++							ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
++							if (ret < 0)
++								return ret;
++						}
++						dapm_update_bits(w);
++						if (w->event_flags & SND_SOC_DAPM_POST_PMU){
++							ret = w->event(w, SND_SOC_DAPM_POST_PMU);
++							if (ret < 0)
++								return ret;
++						}
++					} else {
++						/* power down event */
++						if (w->event_flags & SND_SOC_DAPM_PRE_PMD) {
++							ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
++							if (ret < 0)
++								return ret;
++						}
++						dapm_update_bits(w);
++						if (w->event_flags & SND_SOC_DAPM_POST_PMD) {
++							ret = w->event(w, SND_SOC_DAPM_POST_PMD);
++							if (ret < 0)
++								return ret;
++						}
++					}
++				} else
++					/* no event handler */
++					dapm_update_bits(w);
++			}
++		}
++	}
++
++	return ret;
++}
++
++#if DAPM_DEBUG
++static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
++{
++	struct snd_soc_dapm_widget *w;
++	struct snd_soc_dapm_path *p = NULL;
++	int in, out;
++
++	printk("DAPM %s %s\n", codec->name, action);
++
++	list_for_each_entry(w, &codec->dapm_widgets, list) {
++
++		/* only display widgets that effect routing */
++		switch (w->id) {
++		case snd_soc_dapm_pre:
++		case snd_soc_dapm_post:
++		case snd_soc_dapm_vmid:
++			continue;
++		case snd_soc_dapm_mux:
++		case snd_soc_dapm_output:
++		case snd_soc_dapm_input:
++		case snd_soc_dapm_switch:
++		case snd_soc_dapm_hp:
++		case snd_soc_dapm_mic:
++		case snd_soc_dapm_spk:
++		case snd_soc_dapm_line:
++		case snd_soc_dapm_micbias:
++		case snd_soc_dapm_dac:
++		case snd_soc_dapm_adc:
++		case snd_soc_dapm_pga:
++		case snd_soc_dapm_mixer:
++			if (w->name) {
++				in = is_connected_input_ep(w);
++				dapm_clear_walk(w->codec);
++				out = is_connected_output_ep(w);
++				dapm_clear_walk(w->codec);
++				printk("%s: %s  in %d out %d\n", w->name,
++					w->power ? "On":"Off",in, out);
++
++				list_for_each_entry(p, &w->sources, list_sink) {
++					if (p->connect)
++						printk(" in  %s %s\n", p->name ? p->name : "static",
++							p->source->name);
++				}
++				list_for_each_entry(p, &w->sinks, list_source) {
++					if (p->connect)
++						printk(" out %s %s\n", p->name ? p->name : "static",
++							p->sink->name);
++				}
++			}
++		break;
++		}
++	}
++}
++#endif
++
++/* test and update the power status of a mux widget */
++static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
++				 struct snd_kcontrol *kcontrol, int mask,
++				 int val, struct soc_enum* e)
++{
++	struct snd_soc_dapm_path *path;
++	int found = 0;
++
++	if (widget->id != snd_soc_dapm_mux)
++		return -ENODEV;
++
++	if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
++		return 0;
++
++	/* find dapm widget path assoc with kcontrol */
++	list_for_each_entry(path, &widget->codec->dapm_paths, list) {
++		if (path->kcontrol != kcontrol)
++			continue;
++
++		if (!path->name || ! e->texts[val])
++			continue;
++
++		found = 1;
++		/* we now need to match the string in the enum to the path */
++		if (!(strcmp(path->name, e->texts[val])))
++			path->connect = 1; /* new connection */
++		else
++			path->connect = 0; /* old connection must be powered down */
++	}
++
++	if (found)
++		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
++
++	return 0;
++}
++
++/* test and update the power status of a mixer widget */
++static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
++				   struct snd_kcontrol *kcontrol, int reg,
++				   int val_mask, int val, int invert)
++{
++	struct snd_soc_dapm_path *path;
++	int found = 0;
++
++	if (widget->id != snd_soc_dapm_mixer)
++		return -ENODEV;
++
++	if (!snd_soc_test_bits(widget->codec, reg, val_mask, val))
++		return 0;
++
++	/* find dapm widget path assoc with kcontrol */
++	list_for_each_entry(path, &widget->codec->dapm_paths, list) {
++		if (path->kcontrol != kcontrol)
++			continue;
++
++		/* found, now check type */
++		found = 1;
++		if (val)
++			/* new connection */
++			path->connect = invert ? 0:1;
++		else
++			/* old connection must be powered down */
++			path->connect = invert ? 1:0;
++		break;
++	}
++
++	if (found)
++		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
++
++	return 0;
++}
++
++/* show dapm widget status in sys fs */
++static ssize_t dapm_widget_show(struct device *dev,
++	struct device_attribute *attr, char *buf)
++{
++	struct snd_soc_device *devdata = dev_get_drvdata(dev);
++	struct snd_soc_codec *codec = devdata->codec;
++	struct snd_soc_dapm_widget *w;
++	int count = 0;
++	char *state = "not set";
++
++	list_for_each_entry(w, &codec->dapm_widgets, list) {
++
++		/* only display widgets that burnm power */
++		switch (w->id) {
++		case snd_soc_dapm_hp:
++		case snd_soc_dapm_mic:
++		case snd_soc_dapm_spk:
++		case snd_soc_dapm_line:
++		case snd_soc_dapm_micbias:
++		case snd_soc_dapm_dac:
++		case snd_soc_dapm_adc:
++		case snd_soc_dapm_pga:
++		case snd_soc_dapm_mixer:
++			if (w->name)
++				count += sprintf(buf + count, "%s: %s\n",
++					w->name, w->power ? "On":"Off");
++		break;
++		default:
++		break;
++		}
++	}
++
++	switch(codec->dapm_state){
++	case SNDRV_CTL_POWER_D0:
++		state = "D0";
++		break;
++	case SNDRV_CTL_POWER_D1:
++		state = "D1";
++		break;
++	case SNDRV_CTL_POWER_D2:
++		state = "D2";
++		break;
++	case SNDRV_CTL_POWER_D3hot:
++		state = "D3hot";
++		break;
++	case SNDRV_CTL_POWER_D3cold:
++		state = "D3cold";
++		break;
++	}
++	count += sprintf(buf + count, "PM State: %s\n", state);
++
++	return count;
++}
++
++static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
++
++int snd_soc_dapm_sys_add(struct device *dev)
++{
++	int ret = 0;
++
++	if (dapm_status)
++		ret = device_create_file(dev, &dev_attr_dapm_widget);
++
++	return ret;
++}
++
++static void snd_soc_dapm_sys_remove(struct device *dev)
++{
++	if (dapm_status)
++		device_remove_file(dev, &dev_attr_dapm_widget);
++}
++
++/* free all dapm widgets and resources */
++static void dapm_free_widgets(struct snd_soc_codec *codec)
++{
++	struct snd_soc_dapm_widget *w, *next_w;
++	struct snd_soc_dapm_path *p, *next_p;
++
++	list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
++		list_del(&w->list);
++		kfree(w);
++	}
++
++	list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
++		list_del(&p->list);
++		kfree(p->long_name);
++		kfree(p);
++	}
++}
++
++/**
++ * snd_soc_dapm_sync_endpoints - scan and power dapm paths
++ * @codec: audio codec
++ *
++ * Walks all dapm audio paths and powers widgets according to their
++ * stream or path usage.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
++{
++	return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
++
++/**
++ * snd_soc_dapm_connect_input - connect dapm widgets
++ * @codec: audio codec
++ * @sink: name of target widget
++ * @control: mixer control name
++ * @source: name of source name
++ *
++ * Connects 2 dapm widgets together via a named audio path. The sink is
++ * the widget receiving the audio signal, whilst the source is the sender
++ * of the audio signal.
++ *
++ * Returns 0 for success else error.
++ */
++int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
++	const char * control, const char *source)
++{
++	struct snd_soc_dapm_path *path;
++	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
++	int ret = 0;
++
++	/* find src and dest widgets */
++	list_for_each_entry(w, &codec->dapm_widgets, list) {
++
++		if (!wsink && !(strcmp(w->name, sink))) {
++			wsink = w;
++			continue;
++		}
++		if (!wsource && !(strcmp(w->name, source))) {
++			wsource = w;
++		}
++	}
++
++	if (wsource == NULL || wsink == NULL)
++		return -ENODEV;
++
++	path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
++	if (!path)
++		return -ENOMEM;
++
++	path->source = wsource;
++	path->sink = wsink;
++	INIT_LIST_HEAD(&path->list);
++	INIT_LIST_HEAD(&path->list_source);
++	INIT_LIST_HEAD(&path->list_sink);
++
++	/* check for external widgets */
++	if (wsink->id == snd_soc_dapm_input) {
++		if (wsource->id == snd_soc_dapm_micbias ||
++			wsource->id == snd_soc_dapm_mic ||
++			wsink->id == snd_soc_dapm_line)
++			wsink->ext = 1;
++	}
++	if (wsource->id == snd_soc_dapm_output) {
++		if (wsink->id == snd_soc_dapm_spk ||
++			wsink->id == snd_soc_dapm_hp ||
++			wsink->id == snd_soc_dapm_line)
++			wsource->ext = 1;
++	}
++
++	/* connect static paths */
++	if (control == NULL) {
++		list_add(&path->list, &codec->dapm_paths);
++		list_add(&path->list_sink, &wsink->sources);
++		list_add(&path->list_source, &wsource->sinks);
++		path->connect = 1;
++		return 0;
++	}
++
++	/* connect dynamic paths */
++	switch(wsink->id) {
++	case snd_soc_dapm_adc:
++	case snd_soc_dapm_dac:
++	case snd_soc_dapm_pga:
++	case snd_soc_dapm_input:
++	case snd_soc_dapm_output:
++	case snd_soc_dapm_micbias:
++	case snd_soc_dapm_vmid:
++	case snd_soc_dapm_pre:
++	case snd_soc_dapm_post:
++		list_add(&path->list, &codec->dapm_paths);
++		list_add(&path->list_sink, &wsink->sources);
++		list_add(&path->list_source, &wsource->sinks);
++		path->connect = 1;
++		return 0;
++	case snd_soc_dapm_mux:
++		ret = dapm_connect_mux(codec, wsource, wsink, path, control,
++			&wsink->kcontrols[0]);
++		if (ret != 0)
++			goto err;
++		break;
++	case snd_soc_dapm_switch:
++	case snd_soc_dapm_mixer:
++		ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
++		if (ret != 0)
++			goto err;
++		break;
++	case snd_soc_dapm_hp:
++	case snd_soc_dapm_mic:
++	case snd_soc_dapm_line:
++	case snd_soc_dapm_spk:
++		list_add(&path->list, &codec->dapm_paths);
++		list_add(&path->list_sink, &wsink->sources);
++		list_add(&path->list_source, &wsource->sinks);
++		path->connect = 0;
++		return 0;
++	}
++	return 0;
++
++err:
++	printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source,
++		control, sink);
++	kfree(path);
++	return ret;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
++
++/**
++ * snd_soc_dapm_new_widgets - add new dapm widgets
++ * @codec: audio codec
++ *
++ * Checks the codec for any new dapm widgets and creates them if found.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
++{
++	struct snd_soc_dapm_widget *w;
++
++	mutex_lock(&codec->mutex);
++	list_for_each_entry(w, &codec->dapm_widgets, list)
++	{
++		if (w->new)
++			continue;
++
++		switch(w->id) {
++		case snd_soc_dapm_switch:
++		case snd_soc_dapm_mixer:
++			dapm_new_mixer(codec, w);
++			break;
++		case snd_soc_dapm_mux:
++			dapm_new_mux(codec, w);
++			break;
++		case snd_soc_dapm_adc:
++		case snd_soc_dapm_dac:
++		case snd_soc_dapm_pga:
++			dapm_new_pga(codec, w);
++			break;
++		case snd_soc_dapm_input:
++		case snd_soc_dapm_output:
++		case snd_soc_dapm_micbias:
++		case snd_soc_dapm_spk:
++		case snd_soc_dapm_hp:
++		case snd_soc_dapm_mic:
++		case snd_soc_dapm_line:
++		case snd_soc_dapm_vmid:
++		case snd_soc_dapm_pre:
++		case snd_soc_dapm_post:
++			break;
++		}
++		w->new = 1;
++	}
++
++	dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
++	mutex_unlock(&codec->mutex);
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
++
++/**
++ * snd_soc_dapm_get_volsw - dapm mixer get callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to get the value of a dapm mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
++	int reg = kcontrol->private_value & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int rshift = (kcontrol->private_value >> 12) & 0x0f;
++	int mask = (kcontrol->private_value >> 16) & 0xff;
++	int invert = (kcontrol->private_value >> 24) & 0x01;
++
++	/* return the saved value if we are powered down */
++	if (widget->id == snd_soc_dapm_pga && !widget->power) {
++		ucontrol->value.integer.value[0] = widget->saved_value;
++		return 0;
++	}
++
++	ucontrol->value.integer.value[0] =
++		(snd_soc_read(widget->codec, reg) >> shift) & mask;
++	if (shift != rshift)
++		ucontrol->value.integer.value[1] =
++			(snd_soc_read(widget->codec, reg) >> rshift) & mask;
++	if (invert) {
++		ucontrol->value.integer.value[0] =
++			mask - ucontrol->value.integer.value[0];
++		if (shift != rshift)
++			ucontrol->value.integer.value[1] =
++				mask - ucontrol->value.integer.value[1];
++	}
++
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
++
++/**
++ * snd_soc_dapm_put_volsw - dapm mixer set callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to set the value of a dapm mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
++	int reg = kcontrol->private_value & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int rshift = (kcontrol->private_value >> 12) & 0x0f;
++	int mask = (kcontrol->private_value >> 16) & 0xff;
++	int invert = (kcontrol->private_value >> 24) & 0x01;
++	unsigned short val, val2, val_mask;
++	int ret;
++
++	val = (ucontrol->value.integer.value[0] & mask);
++
++	if (invert)
++		val = mask - val;
++	val_mask = mask << shift;
++	val = val << shift;
++	if (shift != rshift) {
++		val2 = (ucontrol->value.integer.value[1] & mask);
++		if (invert)
++			val2 = mask - val2;
++		val_mask |= mask << rshift;
++		val |= val2 << rshift;
++	}
++
++	mutex_lock(&widget->codec->mutex);
++	widget->value = val;
++
++	/* save volume value if the widget is powered down */
++	if (widget->id == snd_soc_dapm_pga && !widget->power) {
++		widget->saved_value = val;
++		mutex_unlock(&widget->codec->mutex);
++		return 1;
++	}
++
++	dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert);
++	if (widget->event) {
++		if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
++			ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
++			if (ret < 0)
++				goto out;
++		}
++		ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
++		if (widget->event_flags & SND_SOC_DAPM_POST_REG)
++			ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
++	} else
++		ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
++
++out:
++	mutex_unlock(&widget->codec->mutex);
++	return ret;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
++
++/**
++ * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to get the value of a dapm enumerated double mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++	unsigned short val, bitmask;
++
++	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
++		;
++	val = snd_soc_read(widget->codec, e->reg);
++	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
++	if (e->shift_l != e->shift_r)
++		ucontrol->value.enumerated.item[1] =
++			(val >> e->shift_r) & (bitmask - 1);
++
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
++
++/**
++ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to set the value of a dapm enumerated double mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++	unsigned short val, mux;
++	unsigned short mask, bitmask;
++	int ret = 0;
++
++	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
++		;
++	if (ucontrol->value.enumerated.item[0] > e->mask - 1)
++		return -EINVAL;
++	mux = ucontrol->value.enumerated.item[0];
++	val = mux << e->shift_l;
++	mask = (bitmask - 1) << e->shift_l;
++	if (e->shift_l != e->shift_r) {
++		if (ucontrol->value.enumerated.item[1] > e->mask - 1)
++			return -EINVAL;
++		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
++		mask |= (bitmask - 1) << e->shift_r;
++	}
++
++	mutex_lock(&widget->codec->mutex);
++	widget->value = val;
++	dapm_mux_update_power(widget, kcontrol, mask, mux, e);
++	if (widget->event) {
++		if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
++			ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
++			if (ret < 0)
++				goto out;
++		}
++		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
++		if (widget->event_flags & SND_SOC_DAPM_POST_REG)
++			ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
++	} else
++		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
++
++out:
++	mutex_unlock(&widget->codec->mutex);
++	return ret;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
++
++/**
++ * snd_soc_dapm_new_control - create new dapm control
++ * @codec: audio codec
++ * @widget: widget template
++ *
++ * Creates a new dapm control based upon the template.
++ *
++ * Returns 0 for success else error.
++ */
++int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
++	const struct snd_soc_dapm_widget *widget)
++{
++	struct snd_soc_dapm_widget *w;
++
++	if ((w = dapm_cnew_widget(widget)) == NULL)
++		return -ENOMEM;
++
++	w->codec = codec;
++	INIT_LIST_HEAD(&w->sources);
++	INIT_LIST_HEAD(&w->sinks);
++	INIT_LIST_HEAD(&w->list);
++	list_add(&w->list, &codec->dapm_widgets);
++
++	/* machine layer set ups unconnected pins and insertions */
++	w->connected = 1;
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
++
++/**
++ * snd_soc_dapm_stream_event - send a stream event to the dapm core
++ * @codec: audio codec
++ * @stream: stream name
++ * @event: stream event
++ *
++ * Sends a stream event to the dapm core. The core then makes any
++ * necessary widget power changes.
++ *
++ * Returns 0 for success else error.
++ */
++int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
++	char *stream, int event)
++{
++	struct snd_soc_dapm_widget *w;
++
++	if (stream == NULL)
++		return 0;
++
++	mutex_lock(&codec->mutex);
++	list_for_each_entry(w, &codec->dapm_widgets, list)
++	{
++		if (!w->sname)
++			continue;
++		dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname,
++			stream, event);
++		if (strstr(w->sname, stream)) {
++			switch(event) {
++			case SND_SOC_DAPM_STREAM_START:
++				w->active = 1;
++				break;
++			case SND_SOC_DAPM_STREAM_STOP:
++				w->active = 0;
++				break;
++			case SND_SOC_DAPM_STREAM_SUSPEND:
++				if (w->active)
++					w->suspend = 1;
++				w->active = 0;
++				break;
++			case SND_SOC_DAPM_STREAM_RESUME:
++				if (w->suspend) {
++					w->active = 1;
++					w->suspend = 0;
++				}
++				break;
++			case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
++				break;
++			case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
++				break;
++			}
++		}
++	}
++	mutex_unlock(&codec->mutex);
++
++	dapm_power_widgets(codec, event);
++	dump_dapm(codec, __FUNCTION__);
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
++
++/**
++ * snd_soc_dapm_set_endpoint - set audio endpoint status
++ * @codec: audio codec
++ * @endpoint: audio signal endpoint (or start point)
++ * @status: point status
++ *
++ * Set audio endpoint status - connected or disconnected.
++ *
++ * Returns 0 for success else error.
++ */
++int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
++	char *endpoint, int status)
++{
++	struct snd_soc_dapm_widget *w;
++
++	list_for_each_entry(w, &codec->dapm_widgets, list) {
++		if (!strcmp(w->name, endpoint)) {
++			w->connected = status;
++		}
++	}
++
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
++
++/**
++ * snd_soc_dapm_free - free dapm resources
++ * @socdev: SoC device
++ *
++ * Free all dapm widgets and resources.
++ */
++void snd_soc_dapm_free(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++
++	snd_soc_dapm_sys_remove(socdev->dev);
++	dapm_free_widgets(codec);
++}
++EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/soc-core.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/soc-core.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,1610 @@
++/*
++ * soc-core.c  --  ALSA SoC Audio Layer
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *         with code, comments and ideas from :-
++ *         Richard Purdie <richard at 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
++ *    12th Aug 2005   Initial version.
++ *    25th Oct 2005   Working Codec, Interface and Platform registration.
++ *
++ *  TODO:
++ *   o Add hw rules to enforce rates, etc.
++ *   o More testing with other codecs/machines.
++ *   o Add more codecs and platforms to ensure good API coverage.
++ *   o Support TDM on PCM and I2S
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/bitops.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>
++
++/* debug */
++#define SOC_DEBUG 1
++#if SOC_DEBUG
++#define dbg(format, arg...) printk(format, ## arg)
++#else
++#define dbg(format, arg...)
++#endif
++
++static DEFINE_MUTEX(pcm_mutex);
++static DEFINE_MUTEX(io_mutex);
++static struct workqueue_struct *soc_workq;
++static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
++
++/* supported sample rates */
++/* ATTENTION: these values depend on the definition in pcm.h! */
++static const unsigned int rates[] = {
++	5512, 8000, 11025, 16000, 22050, 32000, 44100,
++	48000, 64000, 88200, 96000, 176400, 192000
++};
++
++/*
++ * This is a timeout to do a DAPM powerdown after a stream is closed().
++ * It can be used to eliminate pops between different playback streams, e.g.
++ * between two audio tracks.
++ */
++static int pmdown_time = 5000;
++module_param(pmdown_time, int, 0);
++MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
++
++#ifdef CONFIG_SND_SOC_AC97_BUS
++/* unregister ac97 codec */
++static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
++{
++	if (codec->ac97->dev.bus)
++		device_unregister(&codec->ac97->dev);
++	return 0;
++}
++
++/* stop no dev release warning */
++static void soc_ac97_device_release(struct device *dev){}
++
++/* register ac97 codec to bus */
++static int soc_ac97_dev_register(struct snd_soc_codec *codec)
++{
++	int err;
++
++	codec->ac97->dev.bus = &ac97_bus_type;
++	codec->ac97->dev.parent = NULL;
++	codec->ac97->dev.release = soc_ac97_device_release;
++
++	snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
++		 codec->card->number, 0, codec->name);
++	err = device_register(&codec->ac97->dev);
++	if (err < 0) {
++		snd_printk(KERN_ERR "Can't register ac97 bus\n");
++		codec->ac97->dev.bus = NULL;
++		return err;
++	}
++	return 0;
++}
++#endif
++
++static inline const char* get_dai_name(int type)
++{
++	switch(type) {
++	case SND_SOC_DAI_AC97:
++		return "AC97";
++	case SND_SOC_DAI_I2S:
++		return "I2S";
++	case SND_SOC_DAI_PCM:
++		return "PCM";
++	}
++	return NULL;
++}
++
++/*
++ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
++ * then initialized and any private data can be allocated. This also calls
++ * startup for the cpu DAI, platform, machine and codec DAI.
++ */
++static int soc_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct snd_soc_dai_link *machine = rtd->dai;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
++	int ret = 0;
++
++	mutex_lock(&pcm_mutex);
++
++	/* startup the audio subsystem */
++	if (cpu_dai->ops.startup) {
++		ret = cpu_dai->ops.startup(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't open interface %s\n",
++				cpu_dai->name);
++			goto out;
++		}
++	}
++
++	if (platform->pcm_ops->open) {
++		ret = platform->pcm_ops->open(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
++			goto platform_err;
++		}
++	}
++
++	if (codec_dai->ops.startup) {
++		ret = codec_dai->ops.startup(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't open codec %s\n",
++				codec_dai->name);
++			goto codec_dai_err;
++		}
++	}
++
++	if (machine->ops && machine->ops->startup) {
++		ret = machine->ops->startup(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
++			goto machine_err;
++		}
++	}
++
++	/* Check that the codec and cpu DAI's are compatible */
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		runtime->hw.rate_min =
++			max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min);
++		runtime->hw.rate_max =
++			min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max);
++		runtime->hw.channels_min =
++			max(codec_dai->playback.channels_min,
++				cpu_dai->playback.channels_min);
++		runtime->hw.channels_max =
++			min(codec_dai->playback.channels_max,
++				cpu_dai->playback.channels_max);
++		runtime->hw.formats =
++			codec_dai->playback.formats & cpu_dai->playback.formats;
++		runtime->hw.rates =
++			codec_dai->playback.rates & cpu_dai->playback.rates;
++	} else {
++		runtime->hw.rate_min =
++			max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min);
++		runtime->hw.rate_max =
++			min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max);
++		runtime->hw.channels_min =
++			max(codec_dai->capture.channels_min,
++				cpu_dai->capture.channels_min);
++		runtime->hw.channels_max =
++			min(codec_dai->capture.channels_max,
++				cpu_dai->capture.channels_max);
++		runtime->hw.formats =
++			codec_dai->capture.formats & cpu_dai->capture.formats;
++		runtime->hw.rates =
++			codec_dai->capture.rates & cpu_dai->capture.rates;
++	}
++
++	snd_pcm_limit_hw_rates(runtime);
++	if (!runtime->hw.rates) {
++		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
++			codec_dai->name, cpu_dai->name);
++		goto machine_err;
++	}
++	if (!runtime->hw.formats) {
++		printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
++			codec_dai->name, cpu_dai->name);
++		goto machine_err;
++	}
++	if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
++		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
++			codec_dai->name, cpu_dai->name);
++		goto machine_err;
++	}
++
++	dbg("asoc: %s <-> %s info:\n",codec_dai->name, cpu_dai->name);
++	dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
++	dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
++		runtime->hw.channels_max);
++	dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
++		runtime->hw.rate_max);
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		cpu_dai->playback.active = codec_dai->playback.active = 1;
++	else
++		cpu_dai->capture.active = codec_dai->capture.active = 1;
++	cpu_dai->active = codec_dai->active = 1;
++	cpu_dai->runtime = runtime;
++	socdev->codec->active++;
++	mutex_unlock(&pcm_mutex);
++	return 0;
++
++machine_err:
++	if (machine->ops && machine->ops->shutdown)
++		machine->ops->shutdown(substream);
++
++codec_dai_err:
++	if (platform->pcm_ops->close)
++		platform->pcm_ops->close(substream);
++
++platform_err:
++	if (cpu_dai->ops.shutdown)
++		cpu_dai->ops.shutdown(substream);
++out:
++	mutex_unlock(&pcm_mutex);
++	return ret;
++}
++
++/*
++ * Power down the audio subsytem pmdown_time msecs after close is called.
++ * This is to ensure there are no pops or clicks in between any music tracks
++ * due to DAPM power cycling.
++ */
++static void close_delayed_work(struct work_struct *work)
++{
++	struct snd_soc_device *socdev =
++		container_of(work, struct snd_soc_device, close_work.work);
++	struct snd_soc_codec *codec = socdev->codec;
++	struct snd_soc_codec_dai *codec_dai;
++	int i;
++
++	mutex_lock(&pcm_mutex);
++	for(i = 0; i < codec->num_dai; i++) {
++		codec_dai = &codec->dai[i];
++
++		dbg("pop wq checking: %s status: %s waiting: %s\n",
++			codec_dai->playback.stream_name,
++			codec_dai->playback.active ? "active" : "inactive",
++			codec_dai->pop_wait ? "yes" : "no");
++
++		/* are we waiting on this codec DAI stream */
++		if (codec_dai->pop_wait == 1) {
++
++			codec_dai->pop_wait = 0;
++			snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,
++				SND_SOC_DAPM_STREAM_STOP);
++
++			/* power down the codec power domain if no longer active */
++			if (codec->active == 0) {
++				dbg("pop wq D3 %s %s\n", codec->name,
++					codec_dai->playback.stream_name);
++		 		if (codec->dapm_event)
++					codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++			}
++		}
++	}
++	mutex_unlock(&pcm_mutex);
++}
++
++/*
++ * Called by ALSA when a PCM substream is closed. Private data can be
++ * freed here. The cpu DAI, codec DAI, machine and platform are also
++ * shutdown.
++ */
++static int soc_codec_close(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_dai_link *machine = rtd->dai;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
++	struct snd_soc_codec *codec = socdev->codec;
++
++	mutex_lock(&pcm_mutex);
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		cpu_dai->playback.active = codec_dai->playback.active = 0;
++	else
++		cpu_dai->capture.active = codec_dai->capture.active = 0;
++
++	if (codec_dai->playback.active == 0 &&
++		codec_dai->capture.active == 0) {
++		cpu_dai->active = codec_dai->active = 0;
++	}
++	socdev->codec->active--;
++
++	if (cpu_dai->ops.shutdown)
++		cpu_dai->ops.shutdown(substream);
++
++	if (codec_dai->ops.shutdown)
++		codec_dai->ops.shutdown(substream);
++
++	if (machine->ops && machine->ops->shutdown)
++		machine->ops->shutdown(substream);
++
++	if (platform->pcm_ops->close)
++		platform->pcm_ops->close(substream);
++	cpu_dai->runtime = NULL;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		/* start delayed pop wq here for playback streams */
++		codec_dai->pop_wait = 1;
++		queue_delayed_work(soc_workq, &socdev->close_work,
++			msecs_to_jiffies(pmdown_time));
++	} else {
++		/* capture streams can be powered down now */
++		snd_soc_dapm_stream_event(codec,
++			codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP);
++
++		if (codec->active == 0 && codec_dai->pop_wait == 0){
++			if (codec->dapm_event)
++				codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++		}
++	}
++
++	mutex_unlock(&pcm_mutex);
++	return 0;
++}
++
++/*
++ * Called by ALSA when the PCM substream is prepared, can set format, sample
++ * rate, etc.  This function is non atomic and can be called multiple times,
++ * it can refer to the runtime info.
++ */
++static int soc_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_dai_link *machine = rtd->dai;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
++	struct snd_soc_codec *codec = socdev->codec;
++	int ret = 0;
++
++	mutex_lock(&pcm_mutex);
++
++	if (machine->ops && machine->ops->prepare) {
++		ret = machine->ops->prepare(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: machine prepare error\n");
++			goto out;
++		}
++	}
++
++	if (platform->pcm_ops->prepare) {
++		ret = platform->pcm_ops->prepare(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: platform prepare error\n");
++			goto out;
++		}
++	}
++
++	if (codec_dai->ops.prepare) {
++		ret = codec_dai->ops.prepare(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: codec DAI prepare error\n");
++			goto out;
++		}
++	}
++
++	if (cpu_dai->ops.prepare) {
++		ret = cpu_dai->ops.prepare(substream);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
++			goto out;
++		}
++	}
++
++	/* we only want to start a DAPM playback stream if we are not waiting
++	 * on an existing one stopping */
++	if (codec_dai->pop_wait) {
++		/* we are waiting for the delayed work to start */
++		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
++				snd_soc_dapm_stream_event(socdev->codec,
++					codec_dai->capture.stream_name,
++					SND_SOC_DAPM_STREAM_START);
++		else {
++			codec_dai->pop_wait = 0;
++			cancel_delayed_work(&socdev->close_work);
++			if (codec_dai->dai_ops.digital_mute)
++				codec_dai->dai_ops.digital_mute(codec_dai, 0);
++		}
++	} else {
++		/* no delayed work - do we need to power up codec */
++		if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
++
++			if (codec->dapm_event)
++				codec->dapm_event(codec, SNDRV_CTL_POWER_D1);
++
++			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++				snd_soc_dapm_stream_event(codec,
++					codec_dai->playback.stream_name,
++					SND_SOC_DAPM_STREAM_START);
++			else
++				snd_soc_dapm_stream_event(codec,
++					codec_dai->capture.stream_name,
++					SND_SOC_DAPM_STREAM_START);
++
++			if (codec->dapm_event)
++				codec->dapm_event(codec, SNDRV_CTL_POWER_D0);
++			if (codec_dai->dai_ops.digital_mute)
++				codec_dai->dai_ops.digital_mute(codec_dai, 0);
++
++		} else {
++			/* codec already powered - power on widgets */
++			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++				snd_soc_dapm_stream_event(codec,
++					codec_dai->playback.stream_name,
++					SND_SOC_DAPM_STREAM_START);
++			else
++				snd_soc_dapm_stream_event(codec,
++					codec_dai->capture.stream_name,
++					SND_SOC_DAPM_STREAM_START);
++			if (codec_dai->dai_ops.digital_mute)
++				codec_dai->dai_ops.digital_mute(codec_dai, 0);
++		}
++	}
++
++out:
++	mutex_unlock(&pcm_mutex);
++	return ret;
++}
++
++/*
++ * Called by ALSA when the hardware params are set by application. This
++ * function can also be called multiple times and can allocate buffers
++ * (using snd_pcm_lib_* ). It's non-atomic.
++ */
++static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_dai_link *machine = rtd->dai;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
++	int ret = 0;
++
++	mutex_lock(&pcm_mutex);
++
++	if (machine->ops && machine->ops->hw_params) {
++		ret = machine->ops->hw_params(substream, params);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: machine hw_params failed\n");
++			goto out;
++		}
++	}
++
++	if (codec_dai->ops.hw_params) {
++		ret = codec_dai->ops.hw_params(substream, params);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
++				codec_dai->name);
++			goto codec_err;
++		}
++	}
++
++	if (cpu_dai->ops.hw_params) {
++		ret = cpu_dai->ops.hw_params(substream, params);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't set interface %s hw params\n",
++				cpu_dai->name);
++			goto interface_err;
++		}
++	}
++
++	if (platform->pcm_ops->hw_params) {
++		ret = platform->pcm_ops->hw_params(substream, params);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't set platform %s hw params\n",
++				platform->name);
++			goto platform_err;
++		}
++	}
++
++out:
++	mutex_unlock(&pcm_mutex);
++	return ret;
++
++platform_err:
++	if (cpu_dai->ops.hw_free)
++		cpu_dai->ops.hw_free(substream);
++
++interface_err:
++	if (codec_dai->ops.hw_free)
++		codec_dai->ops.hw_free(substream);
++
++codec_err:
++	if(machine->ops && machine->ops->hw_free)
++		machine->ops->hw_free(substream);
++
++	mutex_unlock(&pcm_mutex);
++	return ret;
++}
++
++/*
++ * Free's resources allocated by hw_params, can be called multiple times
++ */
++static int soc_pcm_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_dai_link *machine = rtd->dai;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
++	struct snd_soc_codec *codec = socdev->codec;
++
++	mutex_lock(&pcm_mutex);
++
++	/* apply codec digital mute */
++	if (!codec->active && codec_dai->dai_ops.digital_mute)
++		codec_dai->dai_ops.digital_mute(codec_dai, 1);
++
++	/* free any machine hw params */
++	if (machine->ops && machine->ops->hw_free)
++		machine->ops->hw_free(substream);
++
++	/* free any DMA resources */
++	if (platform->pcm_ops->hw_free)
++		platform->pcm_ops->hw_free(substream);
++
++	/* now free hw params for the DAI's  */
++	if (codec_dai->ops.hw_free)
++		codec_dai->ops.hw_free(substream);
++
++	if (cpu_dai->ops.hw_free)
++		cpu_dai->ops.hw_free(substream);
++
++	mutex_unlock(&pcm_mutex);
++	return 0;
++}
++
++static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_dai_link *machine = rtd->dai;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
++	int ret;
++
++	if (codec_dai->ops.trigger) {
++		ret = codec_dai->ops.trigger(substream, cmd);
++		if (ret < 0)
++			return ret;
++	}
++
++	if (platform->pcm_ops->trigger) {
++		ret = platform->pcm_ops->trigger(substream, cmd);
++		if (ret < 0)
++			return ret;
++	}
++
++	if (cpu_dai->ops.trigger) {
++		ret = cpu_dai->ops.trigger(substream, cmd);
++		if (ret < 0)
++			return ret;
++	}
++	return 0;
++}
++
++/* ASoC PCM operations */
++static struct snd_pcm_ops soc_pcm_ops = {
++	.open		= soc_pcm_open,
++	.close		= soc_codec_close,
++	.hw_params	= soc_pcm_hw_params,
++	.hw_free	= soc_pcm_hw_free,
++	.prepare	= soc_pcm_prepare,
++	.trigger	= soc_pcm_trigger,
++};
++
++#ifdef CONFIG_PM
++/* powers down audio subsystem for suspend */
++static int soc_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ 	struct snd_soc_machine *machine = socdev->machine;
++ 	struct snd_soc_platform *platform = socdev->platform;
++ 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
++	struct snd_soc_codec *codec = socdev->codec;
++	int i;
++
++	/* mute any active DAC's */
++	for(i = 0; i < machine->num_links; i++) {
++		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
++		if (dai->dai_ops.digital_mute && dai->playback.active)
++			dai->dai_ops.digital_mute(dai, 1);
++	}
++
++	if (machine->suspend_pre)
++		machine->suspend_pre(pdev, state);
++
++	for(i = 0; i < machine->num_links; i++) {
++		struct snd_soc_cpu_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
++			cpu_dai->suspend(pdev, cpu_dai);
++		if (platform->suspend)
++			platform->suspend(pdev, cpu_dai);
++	}
++
++	/* close any waiting streams and save state */
++	flush_workqueue(soc_workq);
++	codec->suspend_dapm_state = codec->dapm_state;
++
++	for(i = 0; i < codec->num_dai; i++) {
++		char *stream = codec->dai[i].playback.stream_name;
++		if (stream != NULL)
++			snd_soc_dapm_stream_event(codec, stream,
++				SND_SOC_DAPM_STREAM_SUSPEND);
++		stream = codec->dai[i].capture.stream_name;
++		if (stream != NULL)
++			snd_soc_dapm_stream_event(codec, stream,
++				SND_SOC_DAPM_STREAM_SUSPEND);
++	}
++
++	if (codec_dev->suspend)
++		codec_dev->suspend(pdev, state);
++
++	for(i = 0; i < machine->num_links; i++) {
++		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
++			cpu_dai->suspend(pdev, cpu_dai);
++	}
++
++	if (machine->suspend_post)
++		machine->suspend_post(pdev, state);
++
++	return 0;
++}
++
++/* powers up audio subsystem after a suspend */
++static int soc_resume(struct platform_device *pdev)
++{
++ 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ 	struct snd_soc_machine *machine = socdev->machine;
++ 	struct snd_soc_platform *platform = socdev->platform;
++ 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
++	struct snd_soc_codec *codec = socdev->codec;
++	int i;
++
++	if (machine->resume_pre)
++		machine->resume_pre(pdev);
++
++	for(i = 0; i < machine->num_links; i++) {
++		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
++			cpu_dai->resume(pdev, cpu_dai);
++	}
++
++	if (codec_dev->resume)
++		codec_dev->resume(pdev);
++
++	for(i = 0; i < codec->num_dai; i++) {
++		char* stream = codec->dai[i].playback.stream_name;
++		if (stream != NULL)
++			snd_soc_dapm_stream_event(codec, stream,
++				SND_SOC_DAPM_STREAM_RESUME);
++		stream = codec->dai[i].capture.stream_name;
++		if (stream != NULL)
++			snd_soc_dapm_stream_event(codec, stream,
++				SND_SOC_DAPM_STREAM_RESUME);
++	}
++
++	/* unmute any active DAC's */
++	for(i = 0; i < machine->num_links; i++) {
++		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
++		if (dai->dai_ops.digital_mute && dai->playback.active)
++			dai->dai_ops.digital_mute(dai, 0);
++	}
++
++	for(i = 0; i < machine->num_links; i++) {
++		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
++			cpu_dai->resume(pdev, cpu_dai);
++		if (platform->resume)
++			platform->resume(pdev, cpu_dai);
++	}
++
++	if (machine->resume_post)
++		machine->resume_post(pdev);
++
++	return 0;
++}
++
++#else
++#define soc_suspend	NULL
++#define soc_resume	NULL
++#endif
++
++/* probes a new socdev */
++static int soc_probe(struct platform_device *pdev)
++{
++	int ret = 0, i;
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_machine *machine = socdev->machine;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
++
++	if (machine->probe) {
++		ret = machine->probe(pdev);
++		if(ret < 0)
++			return ret;
++	}
++
++	for (i = 0; i < machine->num_links; i++) {
++		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->probe) {
++			ret = cpu_dai->probe(pdev);
++			if(ret < 0)
++				goto cpu_dai_err;
++		}
++	}
++
++	if (codec_dev->probe) {
++		ret = codec_dev->probe(pdev);
++		if(ret < 0)
++			goto cpu_dai_err;
++	}
++
++	if (platform->probe) {
++		ret = platform->probe(pdev);
++		if(ret < 0)
++			goto platform_err;
++	}
++
++	/* DAPM stream work */
++	soc_workq = create_workqueue("kdapm");
++	if (soc_workq == NULL)
++		goto work_err;
++	INIT_DELAYED_WORK(&socdev->close_work, close_delayed_work);
++	return 0;
++
++work_err:
++	if (platform->remove)
++		platform->remove(pdev);
++
++platform_err:
++	if (codec_dev->remove)
++		codec_dev->remove(pdev);
++
++cpu_dai_err:
++	for (i--; i > 0; i--) {
++		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->remove)
++			cpu_dai->remove(pdev);
++	}
++
++	if (machine->remove)
++		machine->remove(pdev);
++
++	return ret;
++}
++
++/* removes a socdev */
++static int soc_remove(struct platform_device *pdev)
++{
++	int i;
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct snd_soc_machine *machine = socdev->machine;
++	struct snd_soc_platform *platform = socdev->platform;
++	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
++
++	if (soc_workq) {
++		cancel_rearming_delayed_workqueue(soc_workq, &socdev->close_work);
++		destroy_workqueue(soc_workq);
++	}
++
++	if (platform->remove)
++		platform->remove(pdev);
++
++	if (codec_dev->remove)
++		codec_dev->remove(pdev);
++
++	for (i = 0; i < machine->num_links; i++) {
++		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
++		if (cpu_dai->remove)
++			cpu_dai->remove(pdev);
++	}
++
++	if (machine->remove)
++		machine->remove(pdev);
++
++	return 0;
++}
++
++/* ASoC platform driver */
++static struct platform_driver soc_driver = {
++	.driver		= {
++		.name		= "soc-audio",
++	},
++	.probe		= soc_probe,
++	.remove		= soc_remove,
++	.suspend	= soc_suspend,
++	.resume		= soc_resume,
++};
++
++/* create a new pcm */
++static int soc_new_pcm(struct snd_soc_device *socdev,
++	struct snd_soc_dai_link *dai_link, int num)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai;
++	struct snd_soc_pcm_runtime *rtd;
++	struct snd_pcm *pcm;
++	char new_name[64];
++	int ret = 0, playback = 0, capture = 0;
++
++	rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
++	if (rtd == NULL)
++		return -ENOMEM;
++
++	rtd->dai = dai_link;
++	rtd->socdev = socdev;
++	codec_dai->codec = socdev->codec;
++
++	/* check client and interface hw capabilities */
++	sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name,
++		get_dai_name(cpu_dai->type), num);
++
++	if (codec_dai->playback.channels_min)
++		playback = 1;
++	if (codec_dai->capture.channels_min)
++		capture = 1;
++
++	ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
++		capture, &pcm);
++	if (ret < 0) {
++		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
++		kfree(rtd);
++		return ret;
++	}
++	pcm->private_data = rtd;
++	soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
++	soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
++	soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
++	soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
++	soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
++	soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
++	soc_pcm_ops.page = socdev->platform->pcm_ops->page;
++
++	if (playback)
++		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
++
++	if (capture)
++		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
++
++	ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
++	if (ret < 0) {
++		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
++		kfree(rtd);
++		return ret;
++	}
++
++	pcm->private_free = socdev->platform->pcm_free;
++	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
++		cpu_dai->name);
++	return ret;
++}
++
++/* codec register dump */
++static ssize_t codec_reg_show(struct device *dev,
++	struct device_attribute *attr, char *buf)
++{
++	struct snd_soc_device *devdata = dev_get_drvdata(dev);
++	struct snd_soc_codec *codec = devdata->codec;
++	int i, step = 1, count = 0;
++
++	if (!codec->reg_cache_size)
++		return 0;
++
++	if (codec->reg_cache_step)
++		step = codec->reg_cache_step;
++
++	count += sprintf(buf, "%s registers\n", codec->name);
++	for(i = 0; i < codec->reg_cache_size; i += step)
++		count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i));
++
++	return count;
++}
++static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
++
++/**
++ * snd_soc_new_ac97_codec - initailise AC97 device
++ * @codec: audio codec
++ * @ops: AC97 bus operations
++ * @num: AC97 codec number
++ *
++ * Initialises AC97 codec resources for use by ad-hoc devices only.
++ */
++int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
++	struct snd_ac97_bus_ops *ops, int num)
++{
++	mutex_lock(&codec->mutex);
++
++	codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
++	if (codec->ac97 == NULL) {
++		mutex_unlock(&codec->mutex);
++		return -ENOMEM;
++	}
++
++	codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
++	if (codec->ac97->bus == NULL) {
++		kfree(codec->ac97);
++		codec->ac97 = NULL;
++		mutex_unlock(&codec->mutex);
++		return -ENOMEM;
++	}
++
++	codec->ac97->bus->ops = ops;
++	codec->ac97->num = num;
++	mutex_unlock(&codec->mutex);
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
++
++/**
++ * snd_soc_free_ac97_codec - free AC97 codec device
++ * @codec: audio codec
++ *
++ * Frees AC97 codec device resources.
++ */
++void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
++{
++	mutex_lock(&codec->mutex);
++	kfree(codec->ac97->bus);
++	kfree(codec->ac97);
++	codec->ac97 = NULL;
++	mutex_unlock(&codec->mutex);
++}
++EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
++
++/**
++ * snd_soc_update_bits - update codec register bits
++ * @codec: audio codec
++ * @reg: codec register
++ * @mask: register mask
++ * @value: new value
++ *
++ * Writes new register value.
++ *
++ * Returns 1 for change else 0.
++ */
++int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
++				unsigned short mask, unsigned short value)
++{
++	int change;
++	unsigned short old, new;
++
++	mutex_lock(&io_mutex);
++	old = snd_soc_read(codec, reg);
++	new = (old & ~mask) | value;
++	change = old != new;
++	if (change)
++		snd_soc_write(codec, reg, new);
++
++	mutex_unlock(&io_mutex);
++	return change;
++}
++EXPORT_SYMBOL_GPL(snd_soc_update_bits);
++
++/**
++ * snd_soc_test_bits - test register for change
++ * @codec: audio codec
++ * @reg: codec register
++ * @mask: register mask
++ * @value: new value
++ *
++ * Tests a register with a new value and checks if the new value is
++ * different from the old value.
++ *
++ * Returns 1 for change else 0.
++ */
++int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
++				unsigned short mask, unsigned short value)
++{
++	int change;
++	unsigned short old, new;
++
++	mutex_lock(&io_mutex);
++	old = snd_soc_read(codec, reg);
++	new = (old & ~mask) | value;
++	change = old != new;
++	mutex_unlock(&io_mutex);
++
++	return change;
++}
++EXPORT_SYMBOL_GPL(snd_soc_test_bits);
++
++/**
++ * snd_soc_get_rate - get int sample rate
++ * @hwpcmrate: the hardware pcm rate
++ *
++ * Returns the audio rate integaer value, else 0.
++ */
++int snd_soc_get_rate(int hwpcmrate)
++{
++	int rate = ffs(hwpcmrate) - 1;
++
++	if (rate > ARRAY_SIZE(rates))
++		return 0;
++	return rates[rate];
++}
++EXPORT_SYMBOL_GPL(snd_soc_get_rate);
++
++/**
++ * snd_soc_new_pcms - create new sound card and pcms
++ * @socdev: the SoC audio device
++ *
++ * Create a new sound card based upon the codec and interface pcms.
++ *
++ * Returns 0 for success, else error.
++ */
++int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	struct snd_soc_machine *machine = socdev->machine;
++	int ret = 0, i;
++
++	mutex_lock(&codec->mutex);
++
++	/* register a sound card */
++	codec->card = snd_card_new(idx, xid, codec->owner, 0);
++	if (!codec->card) {
++		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
++			codec->name);
++		mutex_unlock(&codec->mutex);
++		return -ENODEV;
++	}
++
++	codec->card->dev = device_create(sound_class, codec->card->parent, 0,
++					  "card%i", codec->card->number);
++	if (IS_ERR(codec->card->dev)){
++			snd_card_free(codec->card);
++			printk(KERN_ERR "asoc: could not create sound dev %s\n",
++				codec->name);
++			return -ENODEV;
++	}
++
++	codec->card->private_data = codec;
++	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
++
++	/* create the pcms */
++	for(i = 0; i < machine->num_links; i++) {
++		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: can't create pcm %s\n",
++				machine->dai_link[i].stream_name);
++			mutex_unlock(&codec->mutex);
++			return ret;
++		}
++	}
++
++	mutex_unlock(&codec->mutex);
++	return ret;
++}
++EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
++
++/**
++ * snd_soc_register_card - register sound card
++ * @socdev: the SoC audio device
++ *
++ * Register a SoC sound card. Also registers an AC97 device if the
++ * codec is AC97 for ad hoc devices.
++ *
++ * Returns 0 for success, else error.
++ */
++int snd_soc_register_card(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	struct snd_soc_machine *machine = socdev->machine;
++	int ret = 0, i, ac97 = 0, err = 0;
++
++	mutex_lock(&codec->mutex);
++	for(i = 0; i < machine->num_links; i++) {
++		if (socdev->machine->dai_link[i].init) {
++			err = socdev->machine->dai_link[i].init(codec);
++			if (err < 0) {
++				printk(KERN_ERR "asoc: failed to init %s\n",
++					socdev->machine->dai_link[i].stream_name);
++				continue;
++			}
++		}
++		if (socdev->machine->dai_link[i].cpu_dai->type == SND_SOC_DAI_AC97)
++			ac97 = 1;
++	}
++	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
++		 "%s", machine->name);
++	snprintf(codec->card->longname, sizeof(codec->card->longname),
++		 "%s (%s)", machine->name, codec->name);
++
++	ret = snd_card_register(codec->card);
++	if (ret < 0) {
++		printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
++				codec->name);
++		goto out;
++	}
++
++#ifdef CONFIG_SND_SOC_AC97_BUS
++	if (ac97) {
++		ret = soc_ac97_dev_register(codec);
++		if (ret < 0) {
++			printk(KERN_ERR "asoc: AC97 device register failed\n");
++			snd_card_free(codec->card);
++			goto out;
++		}
++	}
++#endif
++
++	err = snd_soc_dapm_sys_add(socdev->dev);
++	if (err < 0)
++		printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
++
++	err = device_create_file(socdev->dev, &dev_attr_codec_reg);
++	if (err < 0)
++		printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n");
++out:
++	mutex_unlock(&codec->mutex);
++	return ret;
++}
++EXPORT_SYMBOL_GPL(snd_soc_register_card);
++
++/**
++ * snd_soc_free_pcms - free sound card and pcms
++ * @socdev: the SoC audio device
++ *
++ * Frees sound card and pcms associated with the socdev.
++ * Also unregister the codec if it is an AC97 device.
++ */
++void snd_soc_free_pcms(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++
++	mutex_lock(&codec->mutex);
++#ifdef CONFIG_SND_SOC_AC97_BUS
++	if (codec->ac97)
++		soc_ac97_dev_unregister(codec);
++#endif
++
++	if (codec->card)
++		snd_card_free(codec->card);
++
++	device_remove_file(socdev->dev, &dev_attr_codec_reg);
++	mutex_unlock(&codec->mutex);
++}
++EXPORT_SYMBOL_GPL(snd_soc_free_pcms);
++
++/**
++ * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
++ * @substream: the pcm substream
++ * @hw: the hardware parameters
++ *
++ * Sets the substream runtime hardware parameters.
++ */
++int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
++	const struct snd_pcm_hardware *hw)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	runtime->hw.info = hw->info;
++	runtime->hw.formats = hw->formats;
++	runtime->hw.period_bytes_min = hw->period_bytes_min;
++	runtime->hw.period_bytes_max = hw->period_bytes_max;
++	runtime->hw.periods_min = hw->periods_min;
++	runtime->hw.periods_max = hw->periods_max;
++	runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
++	runtime->hw.fifo_size = hw->fifo_size;
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
++
++/**
++ * snd_soc_cnew - create new control
++ * @_template: control template
++ * @data: control private data
++ * @lnng_name: control long name
++ *
++ * Create a new mixer control from a template control.
++ *
++ * Returns 0 for success, else error.
++ */
++struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
++	void *data, char *long_name)
++{
++	struct snd_kcontrol_new template;
++
++	memcpy(&template, _template, sizeof(template));
++	if (long_name)
++		template.name = long_name;
++	template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
++	template.index = 0;
++
++	return snd_ctl_new1(&template, data);
++}
++EXPORT_SYMBOL_GPL(snd_soc_cnew);
++
++/**
++ * snd_soc_info_enum_double - enumerated double mixer info callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to provide information about a double enumerated
++ * mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_info *uinfo)
++{
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++
++	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
++	uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
++	uinfo->value.enumerated.items = e->mask;
++
++	if (uinfo->value.enumerated.item > e->mask - 1)
++		uinfo->value.enumerated.item = e->mask - 1;
++	strcpy(uinfo->value.enumerated.name,
++		e->texts[uinfo->value.enumerated.item]);
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
++
++/**
++ * snd_soc_get_enum_double - enumerated double mixer get callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to get the value of a double enumerated mixer.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++	unsigned short val, bitmask;
++
++	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
++		;
++	val = snd_soc_read(codec, e->reg);
++	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
++	if (e->shift_l != e->shift_r)
++		ucontrol->value.enumerated.item[1] =
++			(val >> e->shift_r) & (bitmask - 1);
++
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
++
++/**
++ * snd_soc_put_enum_double - enumerated double mixer put callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to set the value of a double enumerated mixer.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++	unsigned short val;
++	unsigned short mask, bitmask;
++
++	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
++		;
++	if (ucontrol->value.enumerated.item[0] > e->mask - 1)
++		return -EINVAL;
++	val = ucontrol->value.enumerated.item[0] << e->shift_l;
++	mask = (bitmask - 1) << e->shift_l;
++	if (e->shift_l != e->shift_r) {
++		if (ucontrol->value.enumerated.item[1] > e->mask - 1)
++			return -EINVAL;
++		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
++		mask |= (bitmask - 1) << e->shift_r;
++	}
++
++	return snd_soc_update_bits(codec, e->reg, mask, val);
++}
++EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
++
++/**
++ * snd_soc_info_enum_ext - external enumerated single mixer info callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to provide information about an external enumerated
++ * single mixer.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_info *uinfo)
++{
++	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
++
++	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
++	uinfo->count = 1;
++	uinfo->value.enumerated.items = e->mask;
++
++	if (uinfo->value.enumerated.item > e->mask - 1)
++		uinfo->value.enumerated.item = e->mask - 1;
++	strcpy(uinfo->value.enumerated.name,
++		e->texts[uinfo->value.enumerated.item]);
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
++
++/**
++ * snd_soc_info_volsw_ext - external single mixer info callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to provide information about a single external mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_info *uinfo)
++{
++	int mask = kcontrol->private_value;
++
++	uinfo->type =
++		mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
++	uinfo->count = 1;
++	uinfo->value.integer.min = 0;
++	uinfo->value.integer.max = mask;
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
++
++/**
++ * snd_soc_info_bool_ext - external single boolean mixer info callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to provide information about a single boolean external mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_info *uinfo)
++{
++	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
++	uinfo->count = 1;
++	uinfo->value.integer.min = 0;
++	uinfo->value.integer.max = 1;
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_info_bool_ext);
++
++/**
++ * snd_soc_info_volsw - single mixer info callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to provide information about a single mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_info *uinfo)
++{
++	int mask = (kcontrol->private_value >> 16) & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int rshift = (kcontrol->private_value >> 12) & 0x0f;
++
++	uinfo->type =
++		mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
++	uinfo->count = shift == rshift ? 1 : 2;
++	uinfo->value.integer.min = 0;
++	uinfo->value.integer.max = mask;
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
++
++/**
++ * snd_soc_get_volsw - single mixer get callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to get the value of a single mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++	int reg = kcontrol->private_value & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int rshift = (kcontrol->private_value >> 12) & 0x0f;
++	int mask = (kcontrol->private_value >> 16) & 0xff;
++	int invert = (kcontrol->private_value >> 24) & 0x01;
++
++	ucontrol->value.integer.value[0] =
++		(snd_soc_read(codec, reg) >> shift) & mask;
++	if (shift != rshift)
++		ucontrol->value.integer.value[1] =
++			(snd_soc_read(codec, reg) >> rshift) & mask;
++	if (invert) {
++		ucontrol->value.integer.value[0] =
++			mask - ucontrol->value.integer.value[0];
++		if (shift != rshift)
++			ucontrol->value.integer.value[1] =
++				mask - ucontrol->value.integer.value[1];
++	}
++
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
++
++/**
++ * snd_soc_put_volsw - single mixer put callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to set the value of a single mixer control.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++	int reg = kcontrol->private_value & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int rshift = (kcontrol->private_value >> 12) & 0x0f;
++	int mask = (kcontrol->private_value >> 16) & 0xff;
++	int invert = (kcontrol->private_value >> 24) & 0x01;
++	int err;
++	unsigned short val, val2, val_mask;
++
++	val = (ucontrol->value.integer.value[0] & mask);
++	if (invert)
++		val = mask - val;
++	val_mask = mask << shift;
++	val = val << shift;
++	if (shift != rshift) {
++		val2 = (ucontrol->value.integer.value[1] & mask);
++		if (invert)
++			val2 = mask - val2;
++		val_mask |= mask << rshift;
++		val |= val2 << rshift;
++	}
++	err = snd_soc_update_bits(codec, reg, val_mask, val);
++	return err;
++}
++EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
++
++/**
++ * snd_soc_info_volsw_2r - double mixer info callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to provide information about a double mixer control that
++ * spans 2 codec registers.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_info *uinfo)
++{
++	int mask = (kcontrol->private_value >> 12) & 0xff;
++
++	uinfo->type =
++		mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
++	uinfo->count = 2;
++	uinfo->value.integer.min = 0;
++	uinfo->value.integer.max = mask;
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
++
++/**
++ * snd_soc_get_volsw_2r - double mixer get callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to get the value of a double mixer control that spans 2 registers.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++	int reg = kcontrol->private_value & 0xff;
++	int reg2 = (kcontrol->private_value >> 24) & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int mask = (kcontrol->private_value >> 12) & 0xff;
++	int invert = (kcontrol->private_value >> 20) & 0x01;
++
++	ucontrol->value.integer.value[0] =
++		(snd_soc_read(codec, reg) >> shift) & mask;
++	ucontrol->value.integer.value[1] =
++		(snd_soc_read(codec, reg2) >> shift) & mask;
++	if (invert) {
++		ucontrol->value.integer.value[0] =
++			mask - ucontrol->value.integer.value[0];
++		ucontrol->value.integer.value[1] =
++			mask - ucontrol->value.integer.value[1];
++	}
++
++	return 0;
++}
++EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
++
++/**
++ * snd_soc_put_volsw_2r - double mixer set callback
++ * @kcontrol: mixer control
++ * @uinfo: control element information
++ *
++ * Callback to set the value of a double mixer control that spans 2 registers.
++ *
++ * Returns 0 for success.
++ */
++int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++	int reg = kcontrol->private_value & 0xff;
++	int reg2 = (kcontrol->private_value >> 24) & 0xff;
++	int shift = (kcontrol->private_value >> 8) & 0x0f;
++	int mask = (kcontrol->private_value >> 12) & 0xff;
++	int invert = (kcontrol->private_value >> 20) & 0x01;
++	int err;
++	unsigned short val, val2, val_mask;
++
++	val_mask = mask << shift;
++	val = (ucontrol->value.integer.value[0] & mask);
++	val2 = (ucontrol->value.integer.value[1] & mask);
++
++	if (invert) {
++		val = mask - val;
++		val2 = mask - val2;
++	}
++
++	val = val << shift;
++	val2 = val2 << shift;
++
++	if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
++		return err;
++
++	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
++	return err;
++}
++EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
++
++static int __devinit snd_soc_init(void)
++{
++	printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
++	return platform_driver_register(&soc_driver);
++}
++
++static void snd_soc_exit(void)
++{
++ 	platform_driver_unregister(&soc_driver);
++}
++
++module_init(snd_soc_init);
++module_exit(snd_soc_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC Core");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/Kconfig
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,24 @@
++menu "SoC Audio for the Atmel AT91"
++
++config SND_AT91_SOC
++	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
++	depends on ARCH_AT91 && SND
++	select SND_PCM
++	help
++	  Say Y or M if you want to add support for codecs attached to
++	  the AT91 SSC interface. You will also need
++	  to select the audio interfaces to support below.
++
++config SND_AT91_SOC_I2S
++	tristate
++
++config SND_AT91_SOC_ETI_B1_WM8731
++	tristate "SoC I2S Audio support for Endrelia ETI-B1 board"
++	depends on SND_AT91_SOC && MACH_ETI_B1
++	select SND_AT91_SOC_I2S
++	select SND_SOC_WM8731
++	help
++	  Say Y if you want to add support for SoC audio on Endrelia
++          ETI-B1 board.
++
++endmenu
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/Makefile
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/Makefile	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,11 @@
++# AT91 Platform Support
++snd-soc-at91-objs := at91-pcm.o
++snd-soc-at91-i2s-objs := at91-i2s.o
++
++obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
++obj-$(CONFIG_SND_AT91_SOC_I2S) += snd-soc-at91-i2s.o
++
++# AT91 Machine Support
++snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
++
++obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,591 @@
++/*
++ * imx-ssi.c  --  SSI driver for Freescale IMX
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  Based on mxc-alsa-mc13783 (C) 2006 Freescale.
++ *
++ *  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 Aug 2006   Initial version.
++ *
++ */
++
++#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/arch/dma.h>
++#include <asm/arch/spba.h>
++#include <asm/arch/clock.h>
++#include <asm/mach-types.h>
++#include <asm/hardware.h>
++
++#include "imx-ssi.h"
++#include "imx31-pcm.h"
++
++static struct mxc_pcm_dma_params imx_ssi1_pcm_stereo_out = {
++	.name			= "SSI1 PCM Stereo out",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = emi_2_per,
++		.watermark_level = SDMA_TXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI1_STX0,
++		.event_id = DMA_REQ_SSI1_TX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static struct mxc_pcm_dma_params imx_ssi1_pcm_stereo_in = {
++	.name			= "SSI1 PCM Stereo in",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = per_2_emi,
++		.watermark_level = SDMA_RXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI1_SRX0,
++		.event_id = DMA_REQ_SSI1_RX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static struct mxc_pcm_dma_params imx_ssi2_pcm_stereo_out = {
++	.name			= "SSI2 PCM Stereo out",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = per_2_emi,
++		.watermark_level = SDMA_TXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI2_STX0,
++		.event_id = DMA_REQ_SSI2_TX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static struct mxc_pcm_dma_params imx_ssi2_pcm_stereo_in = {
++	.name			= "SSI2 PCM Stereo in",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = per_2_emi,
++		.watermark_level = SDMA_RXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI2_SRX0,
++		.event_id = DMA_REQ_SSI2_RX1,
++		.peripheral_type = SSI,
++	},
++};
++
++/*
++ * SSI system clock configuration.
++ */
++static int imx_ssi_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
++	int clk_id, unsigned int freq, int dir)
++{
++	u32 scr;
++
++	if (cpu_dai->id == IMX_DAI_SSI1)
++		scr = __raw_readw(SSI1_SCR);
++	else
++		scr = __raw_readw(SSI2_SCR);
++
++	switch (clk_id) {
++	case IMX_SSP_SYS_CLK:
++		if (dir == SND_SOC_CLOCK_OUT)
++			scr |= SSI_SCR_SYS_CLK_EN;
++		else
++			scr &= ~SSI_SCR_SYS_CLK_EN;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	if (cpu_dai->id == IMX_DAI_SSI1)
++		__raw_writew(scr, SSI1_SCR);
++	else
++		__raw_writew(scr, SSI2_SCR);
++
++	return 0;
++}
++
++/*
++ * SSI Clock dividers
++ */
++static int imx_ssi_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
++	int div_id, int div)
++{
++	u32 stccr;
++
++	if (cpu_dai->id == IMX_DAI_SSI1)
++		stccr = __raw_readw(SSI1_STCCR);
++	else
++		stccr = __raw_readw(SSI2_STCCR);
++
++	switch (div_id) {
++	case IMX_SSI_DIV_2:
++		stccr &= ~SSI_STCCR_DIV2;
++		stccr |= div;
++		break;
++	case IMX_SSI_DIV_PSR:
++		stccr &= ~SSI_STCCR_PSR;
++		stccr |= div;
++		break;
++	case IMX_SSI_DIV_PM:
++		stccr &= ~0xff;
++		stccr |= SSI_STCCR_PM(div);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	if (cpu_dai->id == IMX_DAI_SSI1)
++		__raw_writew(stccr, SSI1_STCCR);
++	else
++		__raw_writew(stccr, SSI2_STCCR);
++
++	return 0;
++}
++
++/*
++ * SSI Network Mode or TDM slots configuration.
++ */
++static int imx_ssi_set_dai_tdm_slot(struct snd_soc_cpu_dai *cpu_dai,
++	unsigned int mask, int slots)
++{
++	u32 stmsk, srmsk, scr, stccr;
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		stmsk = __raw_readw(SSI1_STMSK);
++		srmsk = __raw_readw(SSI1_SRMSK);
++		scr = __raw_readw(SSI1_SCR);
++		stccr = __raw_readw(SSI1_STCCR);
++	} else {
++		stmsk = __raw_readw(SSI2_STMSK);
++		srmsk = __raw_readw(SSI2_SRMSK);
++		scr = __raw_readw(SSI2_SCR);
++		stccr = __raw_readw(SSI2_STCCR);
++	}
++
++	stmsk = srmsk = mask;
++	scr |= SSI_SCR_NET;
++	stccr &= ~0x1f00;
++	stccr |= SSI_STCCR_DC(slots);
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		__raw_writew(stmsk, SSI1_STMSK);
++		__raw_writew(srmsk, SSI1_SRMSK);
++		__raw_writew(scr, SSI1_SCR);
++		__raw_writew(stccr, SSI1_STCCR);
++	} else {
++		__raw_writew(stmsk, SSI2_STMSK);
++		__raw_writew(srmsk, SSI2_SRMSK);
++		__raw_writew(scr, SSI2_SCR);
++		__raw_writew(stccr, SSI2_STCCR);
++	}
++
++	return 0;
++}
++
++/*
++ * SSI DAI format configuration.
++ */
++static int imx_ssi_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int fmt)
++{
++	u32 stcr = 0, srcr = 0;
++
++	/* DAI mode */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		stcr |= SSI_STCR_TSCKP | SSI_STCR_TFSI |
++			SSI_STCR_TEFS | SSI_STCR_TXBIT0;
++		srcr |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI |
++			SSI_SRCR_REFS | SSI_SRCR_RXBIT0;
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		stcr |= SSI_STCR_TSCKP | SSI_STCR_TFSI | SSI_STCR_TXBIT0;
++		srcr |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI | SSI_SRCR_RXBIT0;
++		break;
++	case SND_SOC_DAIFMT_DSP_B:
++		stcr |= SSI_STCR_TEFS; // data 1 bit after sync
++		srcr |= SSI_SRCR_REFS; // data 1 bit after sync
++	case SND_SOC_DAIFMT_DSP_A:
++		stcr |= SSI_STCR_TFSL; // frame is 1 bclk long
++		srcr |= SSI_SRCR_RFSL; // frame is 1 bclk long
++
++		/* DAI clock inversion */
++		switch(fmt & SND_SOC_DAIFMT_INV_MASK) {
++		case SND_SOC_DAIFMT_IB_IF:
++			stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
++			srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP;
++			break;
++		case SND_SOC_DAIFMT_IB_NF:
++			stcr |= SSI_STCR_TSCKP;
++			srcr |= SSI_SRCR_RSCKP;
++			break;
++		case SND_SOC_DAIFMT_NB_IF:
++			stcr |= SSI_STCR_TFSI;
++			srcr |= SSI_SRCR_RFSI;
++			break;
++		}
++		break;
++	}
++
++	/* DAI clock master masks */
++	switch(fmt & SND_SOC_DAIFMT_CLOCK_MASK){
++	case SND_SOC_DAIFMT_CBM_CFM:
++		stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR;
++		srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFM:
++		stcr |= SSI_STCR_TFDIR;
++		srcr |= SSI_SRCR_RFDIR;
++		break;
++	case SND_SOC_DAIFMT_CBM_CFS:
++		stcr |= SSI_STCR_TXDIR;
++		srcr |= SSI_SRCR_RXDIR;
++		break;
++	}
++
++	/* async */
++	//if (rtd->cpu_dai->flags & SND_SOC_DAI_ASYNC)
++	//	SSI1_SCR |= SSI_SCR_SYN;
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		__raw_writew(stcr, SSI1_STCR);
++		__raw_writew(0, SSI1_STCCR);
++		__raw_writew(srcr, SSI1_SRCR);
++		__raw_writew(0, SSI1_SRCCR);
++	} else {
++		__raw_writew(stcr, SSI2_STCR);
++		__raw_writew(0, SSI2_STCCR);
++		__raw_writew(srcr, SSI2_SRCR);
++		__raw_writew(0, SSI2_SRCCR);
++	}
++
++	return 0;
++}
++
++static int imx_ssi_set_dai_tristate(struct snd_soc_cpu_dai *cpu_dai,
++	int tristate)
++{
++	// via GPIO ??
++	return 0;
++}
++
++static int imx_ssi_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (cpu_dai->id == IMX_DAI_SSI1)
++		mxc_clks_enable(SSI1_BAUD);
++	else
++		mxc_clks_enable(SSI2_BAUD);
++	return 0;
++}
++
++static int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	u32 stccr, stcr;
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		stccr = __raw_readw(SSI1_STCCR) & 0x600ff;
++		stcr = __raw_readw(SSI1_STCR);
++	} else {
++		stccr = __raw_readw(SSI2_STCCR) & 0x600ff;
++		stcr = __raw_readw(SSI2_STCR);
++	}
++
++	/* DAI data (word) size */
++	switch(params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		stccr |= SSI_STCCR_WL(16);
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		stccr |= SSI_STCCR_WL(20);
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		stccr |= SSI_STCCR_WL(24);
++		break;
++	}
++
++	/* TDM - todo, only fifo 0 atm */
++	stcr |= SSI_STCR_TFEN0;
++	stccr |= SSI_STCCR_DC(params_channels(params));
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		__raw_writew(stcr, SSI1_STCR);
++		__raw_writew(stccr, SSI1_STCCR);
++	} else {
++		__raw_writew(stcr, SSI2_STCR);
++		__raw_writew(stccr, SSI2_STCCR);
++	}
++
++	return 0;
++}
++
++static int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	u32 srccr, srcr;
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		srccr = __raw_readw(SSI1_SRCCR) & 0x600ff;
++		srcr = __raw_readw(SSI1_SRCR);
++	} else {
++		srccr = __raw_readw(SSI2_SRCCR) & 0x600ff;
++		srcr = __raw_readw(SSI2_SRCR);
++	}
++
++	/* DAI data (word) size */
++	switch(params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		srccr |= SSI_SRCCR_WL(16);
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		srccr |= SSI_SRCCR_WL(20);
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		srccr |= SSI_SRCCR_WL(24);
++		break;
++	}
++
++	/* TDM - todo, only fifo 0 atm */
++	srcr |= SSI_SRCR_RFEN0;
++	srccr |= SSI_SRCCR_DC(params_channels(params));
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		__raw_writew(srcr, SSI1_SRCR);
++		__raw_writew(srccr, SSI1_SRCCR);
++	} else {
++		__raw_writew(srcr, SSI2_SRCR);
++		__raw_writew(srccr, SSI2_SRCCR);
++	}
++	return 0;
++}
++
++static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	/* Tx/Rx config */
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		if (cpu_dai->id == IMX_DAI_SSI1)
++			cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out;
++		else
++			cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out;
++		return imx_ssi_hw_tx_params(substream, params);
++	} else {
++		if (cpu_dai->id == IMX_DAI_SSI1)
++			cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in;
++		else
++			cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in;
++		return imx_ssi_hw_rx_params(substream, params);
++	}
++}
++
++static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	u32 scr, sier;
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		scr = __raw_readw(SSI1_SCR) & 0x600ff;
++		sier = __raw_readw(SSI1_SIER);
++	} else {
++		scr = __raw_readw(SSI2_SCR) & 0x600ff;
++		sier = __raw_readw(SSI2_SIER);
++	}
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++			scr |= SSI_SCR_TE;
++			sier |= SSI_SIER_TDMAE;
++		} else {
++			scr |= SSI_SCR_RE;
++			sier |= SSI_SIER_RDMAE;
++		}
++		scr |= SSI_SCR_SSIEN;
++		break;
++	case SNDRV_PCM_TRIGGER_RESUME:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			scr |= SSI_SCR_TE;
++		else
++			scr |= SSI_SCR_RE;
++		scr |= SSI_SCR_SSIEN;
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			sier |= SSI_SIER_TDMAE;
++		else
++			sier |= SSI_SIER_RDMAE;
++	break;
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++		scr &= ~SSI_SCR_SSIEN;
++	case SNDRV_PCM_TRIGGER_STOP:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			scr &= ~SSI_SCR_TE;
++		else
++			scr &= ~SSI_SCR_RE;
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++			sier &= ~SSI_SIER_TDMAE;
++		else
++			sier &= ~SSI_SIER_TDMAE;
++	break;
++	default:
++		return -EINVAL;
++	}
++
++	if (cpu_dai->id == IMX_DAI_SSI1) {
++		__raw_writew(scr, SSI1_SCR);
++		__raw_writew(sier, SSI1_SIER);
++	} else {
++		__raw_writew(scr, SSI2_SCR);
++		__raw_writew(sier, SSI2_SIER);
++	}
++
++	return 0;
++}
++
++static void imx_ssi_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	/* shutdown SSI */
++	if (!cpu_dai->active) {
++		if(cpu_dai->id == IMX_DAI_SSI1) {
++			__raw_writew(__raw_readw(SSI1_SCR) & ~SSI_SCR_SSIEN, SSI1_SCR);
++			mxc_clks_disable(SSI1_BAUD);
++		} else {
++			__raw_writew(__raw_readw(SSI2_SCR) & ~SSI_SCR_SSIEN, SSI2_SCR);
++			mxc_clks_disable(SSI2_BAUD);
++		}
++	}
++}
++
++#ifdef CONFIG_PM
++static int imx_ssi_suspend(struct platform_device *dev,
++	struct snd_soc_cpu_dai *dai)
++{
++	if(!dai->active)
++		return 0;
++
++	// do we need to disable any clocks
++
++	return 0;
++}
++
++static int imx_ssi_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++	if(!dai->active)
++		return 0;
++
++	// do we need to enable any clocks
++	return 0;
++}
++
++#else
++#define imx_ssi_suspend	NULL
++#define imx_ssi_resume	NULL
++#endif
++
++#define IMX_SSI_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 IMX_SSI_BITS \
++	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_cpu_dai imx_ssi_pcm_dai[] = {
++{
++	.name = "imx-i2s-1",
++	.id = IMX_DAI_SSI1,
++	.type = SND_SOC_DAI_I2S,
++	.suspend = imx_ssi_suspend,
++	.resume = imx_ssi_resume,
++	.playback = {
++		.channels_min = 1,
++		.channels_max = 2,
++		.formats = IMX_SSI_BITS,
++		.rates = IMX_SSI_RATES,},
++	.capture = {
++		.channels_min = 1,
++		.channels_max = 2,
++		.formats = IMX_SSI_BITS,
++		.rates = IMX_SSI_RATES,},
++	.ops = {
++		.startup = imx_ssi_startup,
++		.shutdown = imx_ssi_shutdown,
++		.trigger = imx_ssi_trigger,
++		.hw_params = imx_ssi_hw_params,},
++	.dai_ops = {
++		.set_sysclk = imx_ssi_set_dai_sysclk,
++		.set_clkdiv = imx_ssi_set_dai_clkdiv,
++		.set_fmt = imx_ssi_set_dai_fmt,
++		.set_tdm_slot = imx_ssi_set_dai_tdm_slot,
++		.set_tristate = imx_ssi_set_dai_tristate,
++	},
++},
++{
++	.name = "imx-i2s-2",
++	.id = IMX_DAI_SSI2,
++	.type = SND_SOC_DAI_I2S,
++	.suspend = imx_ssi_suspend,
++	.resume = imx_ssi_resume,
++	.playback = {
++		.channels_min = 1,
++		.channels_max = 2,
++		.formats = IMX_SSI_BITS,
++		.rates = IMX_SSI_RATES,},
++	.capture = {
++		.channels_min = 1,
++		.channels_max = 2,
++		.formats = IMX_SSI_BITS,
++		.rates = IMX_SSI_RATES,},
++	.ops = {
++		.startup = imx_ssi_startup,
++		.shutdown = imx_ssi_shutdown,
++		.trigger = imx_ssi_trigger,
++		.hw_params = imx_ssi_hw_params,},
++	.dai_ops = {
++		.set_sysclk = imx_ssi_set_dai_sysclk,
++		.set_clkdiv = imx_ssi_set_dai_clkdiv,
++		.set_fmt = imx_ssi_set_dai_fmt,
++		.set_tdm_slot = imx_ssi_set_dai_tdm_slot,
++		.set_tristate = imx_ssi_set_dai_tristate,
++	},
++},};
++EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("i.MX ASoC I2S driver");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/Kconfig
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,31 @@
++menu "SoC Audio for the Freescale i.MX"
++
++config SND_MXC_SOC
++	tristate "SoC Audio for the Freescale i.MX CPU"
++	depends on ARCH_MXC && SND
++	select SND_PCM
++	help
++	  Say Y or M if you want to add support for codecs attached to
++	  the MXC AC97, I2S or SSP interface. You will also need
++	  to select the audio interfaces to support below.
++
++config SND_MXC_AC97
++	tristate
++	select SND_AC97_CODEC
++
++config SND_MXC_SOC_AC97
++	tristate
++	select AC97_BUS
++
++config SND_MXC_SOC_SSI
++	tristate
++
++config SND_SOC_MX31ADS_WM8753
++	tristate "SoC Audio support for MX31 - WM8753"
++	depends on SND_MXC_SOC && ARCH_MX3
++	select SND_MXC_SOC_SSI
++	help
++	  Say Y if you want to add support for SoC audio on MX31ADS
++	  with the WM8753.
++
++endmenu
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/Makefile
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/Makefile	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,18 @@
++# i.MX Platform Support
++snd-soc-imx21-objs := imx21-pcm.o
++snd-soc-imx31-objs := imx31-pcm.o
++snd-soc-imx-ac97-objs := imx-ac97.o
++snd-soc-imx-ssi-objs := imx-ssi.o
++
++obj-$(CONFIG_SND_MXC_SOC) += snd-soc-imx31.o
++obj-$(CONFIG_SND_MXC_SOC_AC97) += snd-soc-imx-ac97.o
++obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-imx-ssi.o
++
++# i.MX Machine Support
++snd-soc-mx31ads-wm8753-objs := mx31ads_wm8753.o
++obj-$(CONFIG_SND_SOC_MX31ADS_WM8753) += snd-soc-mx31ads-wm8753.o
++snd-soc-mx21ads-wm8753-objs := mx21ads_wm8753.o
++obj-$(CONFIG_SND_SOC_MX21ADS_WM8753) += snd-soc-mx21ads-wm8753.o
++snd-soc-mx21ads-wm8731-objs := mx21ads_wm8731.o
++obj-$(CONFIG_SND_SOC_MX21ADS_WM8731) += snd-soc-mx21ads-wm8731.o
++
+Index: linux-2.6.17.14-fic4.test/sound/Makefile
+===================================================================
+--- linux-2.6.17.14-fic4.test.orig/sound/Makefile	2006-10-13 20:55:04.000000000 +0200
++++ linux-2.6.17.14-fic4.test/sound/Makefile	2007-01-24 12:20:33.000000000 +0100
+@@ -4,7 +4,7 @@
+ obj-$(CONFIG_SOUND) += soundcore.o
+ obj-$(CONFIG_SOUND_PRIME) += oss/
+ obj-$(CONFIG_DMASOUND) += oss/
+-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/
++obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
+ 
+ ifeq ($(CONFIG_SND),y)
+   obj-y += last.o
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,708 @@
++/*
++ * wm8711.c  --  WM8711 ALSA SoC Audio driver
++ *
++ * Copyright 2006 Wolfson Microelectronics
++ *
++ * Author: Mike Arthur <linux at wolfsonmicro.com>
++ *
++ * Based on wm8731.c by Richard Purdie
++ *
++ * 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 "wm8711.h"
++
++#define AUDIO_NAME "wm8711"
++#define WM8711_VERSION "0.3"
++
++/*
++ * Debug
++ */
++
++#define WM8711_DEBUG 0
++
++#ifdef WM8711_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_wm8711;
++
++/* codec private data */
++struct wm8711_priv {
++	unsigned int sysclk;
++};
++
++/*
++ * wm8711 register cache
++ * We can't read the WM8711 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 wm8711_reg[WM8711_CACHEREGNUM] = {
++    0x0079, 0x0079, 0x000a, 0x0008,
++    0x009f, 0x000a, 0x0000, 0x0000
++};
++
++/*
++ * read wm8711 register cache
++ */
++static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec * codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg == WM8711_RESET)
++		return 0;
++	if (reg >= WM8711_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write wm8711 register cache
++ */
++static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= WM8711_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the WM8711 register space
++ */
++static int wm8711_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;
++
++	wm8711_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -EIO;
++}
++
++#define wm8711_reset(c)	wm8711_write(c, WM8711_RESET, 0)
++
++static const struct snd_kcontrol_new wm8711_snd_controls[] = {
++
++SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V,
++	0, 127, 0),
++SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V,
++	7, 1, 0),
++
++};
++
++/* add non dapm controls */
++static int wm8711_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8711_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* Output Mixer */
++static const snd_kcontrol_new_t wm8711_output_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0),
++SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0),
++};
++
++static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = {
++SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1,
++	&wm8711_output_mixer_controls[0],
++	ARRAY_SIZE(wm8711_output_mixer_controls)),
++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1),
++SND_SOC_DAPM_OUTPUT("LOUT"),
++SND_SOC_DAPM_OUTPUT("LHPOUT"),
++SND_SOC_DAPM_OUTPUT("ROUT"),
++SND_SOC_DAPM_OUTPUT("RHPOUT"),
++};
++
++static const char *intercon[][3] = {
++	/* output mixer */
++	{"Output Mixer", "Line Bypass Switch", "Line Input"},
++	{"Output Mixer", "HiFi Playback Switch", "DAC"},
++
++	/* outputs */
++	{"RHPOUT", NULL, "Output Mixer"},
++	{"ROUT", NULL, "Output Mixer"},
++	{"LHPOUT", NULL, "Output Mixer"},
++	{"LOUT", NULL, "Output Mixer"},
++
++	/* terminator */
++	{NULL, NULL, NULL},
++};
++
++static int wm8711_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(wm8711_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &wm8711_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;
++}
++
++static int wm8711_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	struct wm8711_priv *wm8711 = codec->private_data;
++	u16 iface = wm8711_read_reg_cache(codec, WM8711_IFACE) & 0xfffc;
++	int i = get_coeff(wm8711->sysclk, snd_soc_get_rate(params_rate(params)));
++	u16 srate = (coeff_div[i].sr << 2) |
++		(coeff_div[i].bosr << 1) | coeff_div[i].usb;
++
++	wm8711_write(codec, WM8711_SRATE, srate);
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0004;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0008;
++		break;
++	}
++
++	wm8711_write(codec, WM8711_IFACE, iface);
++	return 0;
++}
++
++static int wm8711_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;
++
++    /* set active */
++    wm8711_write(codec, WM8711_ACTIVE, 0x0001);
++    return 0;
++}
++
++static void wm8711_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);
++        wm8711_write(codec, WM8711_ACTIVE, 0x0);
++    }
++}
++
++static int wm8711_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++    u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7;
++
++    if (mute)
++        wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8);
++    else
++        wm8711_write(codec, WM8711_APDIGI, mute_reg);
++
++	return 0;
++}
++
++static int wm8711_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	struct wm8711_priv *wm8711 = codec->private_data;
++
++	switch (freq) {
++	case 11289600:
++	case 12000000:
++	case 12288000:
++	case 16934400:
++	case 18432000:
++		wm8711->sysclk = freq;
++		return 0;
++	}
++	return -EINVAL;
++}
++
++static int wm8711_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = 0;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		iface |= 0x0040;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	/* set iface */
++	wm8711_write(codec, WM8711_IFACE, iface);
++	return 0;
++}
++
++
++static int wm8711_dapm_event(struct snd_soc_codec *codec, int event)
++{
++	u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f;
++
++	switch (event) {
++	case SNDRV_CTL_POWER_D0: /* full On */
++		/* vref/mid, osc on, dac unmute */
++		wm8711_write(codec, WM8711_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, */
++		wm8711_write(codec, WM8711_PWR, reg | 0x0040);
++		break;
++	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++		/* everything off, dac mute, inactive */
++		wm8711_write(codec, WM8711_ACTIVE, 0x0);
++		wm8711_write(codec, WM8711_PWR, 0xffff);
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define WM8711_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
++		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
++
++#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_codec_dai wm8711_dai = {
++	.name = "WM8711",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8711_RATES,
++		.formats = WM8711_FORMATS,},
++	.ops = {
++		.prepare = wm8711_pcm_prepare,
++		.hw_params = wm8711_hw_params,
++		.shutdown = wm8711_shutdown,
++	},
++	.dai_ops = {
++		.digital_mute = wm8711_mute,
++		.set_sysclk = wm8711_set_dai_sysclk,
++		.set_fmt = wm8711_set_dai_fmt,
++	},
++};
++EXPORT_SYMBOL_GPL(wm8711_dai);
++
++static int wm8711_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;
++
++	wm8711_write(codec, WM8711_ACTIVE, 0x0);
++	wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int wm8711_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(wm8711_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8711_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the WM8711 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8711_init(struct snd_soc_device* socdev)
++{
++	struct snd_soc_codec* codec = socdev->codec;
++	int reg, ret = 0;
++
++	codec->name = "WM8711";
++	codec->owner = THIS_MODULE;
++	codec->read = wm8711_read_reg_cache;
++	codec->write = wm8711_write;
++	codec->dapm_event = wm8711_dapm_event;
++	codec->dai = &wm8711_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(wm8711_reg);
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8711_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, wm8711_reg,
++		sizeof(u16) * ARRAY_SIZE(wm8711_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8711_reg);
++
++	wm8711_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 */
++	wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++	/* set the update bits */
++	reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V);
++	wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100);
++	reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V);
++	wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100);
++
++	wm8711_add_controls(codec);
++	wm8711_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 *wm8711_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8711 2 wire address is determined by GPIO5
++ * state during powerup.
++ *    low  = 0x1a
++ *    high = 0x1b
++ */
++#define I2C_DRIVERID_WM8711 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 wm8711_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 wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++	struct snd_soc_device *socdev = wm8711_socdev;
++	struct wm8711_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 = wm8711_init(socdev);
++    if (ret < 0) {
++        err("failed to initialise WM8711\n");
++        goto err;
++    }
++    return ret;
++
++err:
++    kfree(codec);
++    kfree(i2c);
++    return ret;
++
++}
++
++static int wm8711_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 wm8711_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, wm8711_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8711_i2c_driver = {
++	.driver = {
++		.name = "WM8711 I2C Codec",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_WM8711,
++	.attach_adapter = wm8711_i2c_attach,
++	.detach_client =  wm8711_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "WM8711",
++	.driver = &wm8711_i2c_driver,
++};
++#endif
++
++static int wm8711_probe(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct wm8711_setup_data *setup;
++	struct snd_soc_codec* codec;
++	struct wm8711_priv *wm8711;
++	int ret = 0;
++
++	info("WM8711 Audio Codec %s", WM8711_VERSION);
++
++	setup = socdev->codec_data;
++	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++	if (codec == NULL)
++		return -ENOMEM;
++
++	wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL);
++	if (wm8711 == NULL) {
++		kfree(codec);
++		return -ENOMEM;
++	}
++
++	codec->private_data = wm8711;
++	socdev->codec = codec;
++	mutex_init(&codec->mutex);
++	INIT_LIST_HEAD(&codec->dapm_widgets);
++	INIT_LIST_HEAD(&codec->dapm_paths);
++
++	wm8711_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(&wm8711_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 wm8711_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)
++		wm8711_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(&wm8711_i2c_driver);
++#endif
++	kfree(codec->private_data);
++	kfree(codec);
++
++	return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8711 = {
++	.probe = 	wm8711_probe,
++	.remove = 	wm8711_remove,
++	.suspend = 	wm8711_suspend,
++	.resume =	wm8711_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711);
++
++MODULE_DESCRIPTION("ASoC WM8711 driver");
++MODULE_AUTHOR("Mike Arthur");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,42 @@
++/*
++ * wm8711.h  --  WM8711 Soc Audio driver
++ *
++ * Copyright 2006 Wolfson Microelectronics
++ *
++ * Author: Mike Arthur <linux at wolfsonmicro.com>
++ *
++ * Based on wm8731.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 _WM8711_H
++#define _WM8711_H
++
++/* WM8711 register space */
++
++#define WM8711_LOUT1V   0x02
++#define WM8711_ROUT1V   0x03
++#define WM8711_APANA    0x04
++#define WM8711_APDIGI   0x05
++#define WM8711_PWR      0x06
++#define WM8711_IFACE    0x07
++#define WM8711_SRATE    0x08
++#define WM8711_ACTIVE   0x09
++#define WM8711_RESET	0x0f
++
++#define WM8711_CACHEREGNUM 	8
++
++#define WM8711_SYSCLK	0
++#define WM8711_DAI		0
++
++struct wm8711_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8711_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8711;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,920 @@
++/*
++ * wm8980.c  --  WM8980 ALSA Soc Audio driver
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ *
++ * Authors:
++ * Mike Arthur      <linux at wolfsonmicro.com>
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License 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 "wm8980.h"
++
++#define AUDIO_NAME "wm8980"
++#define WM8980_VERSION "0.3"
++
++/*
++ * Debug
++ */
++
++#define WM8980_DEBUG 0
++
++#ifdef WM8980_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_wm8980;
++
++/*
++ * wm8980 register cache
++ * We can't read the WM8980 register space when we are
++ * using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8980_reg[WM8980_CACHEREGNUM] = {
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0050, 0x0000, 0x0140, 0x0000,
++    0x0000, 0x0000, 0x0000, 0x00ff,
++    0x00ff, 0x0000, 0x0100, 0x00ff,
++    0x00ff, 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,
++    0x0033, 0x0010, 0x0010, 0x0100,
++    0x0100, 0x0002, 0x0001, 0x0001,
++    0x0039, 0x0039, 0x0039, 0x0039,
++    0x0001, 0x0001,
++};
++
++/*
++ * read wm8980 register cache
++ */
++static inline unsigned int wm8980_read_reg_cache(struct snd_soc_codec  *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg == WM8980_RESET)
++		return 0;
++	if (reg >= WM8980_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write wm8980 register cache
++ */
++static inline void wm8980_write_reg_cache(struct snd_soc_codec  *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= WM8980_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the WM8980 register space
++ */
++static int wm8980_write(struct snd_soc_codec  *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D9 WM8980 register offset
++	 *   D8...D0 register data
++	 */
++	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++	data[1] = value & 0x00ff;
++
++	wm8980_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -1;
++}
++
++#define wm8980_reset(c)	wm8980_write(c, WM8980_RESET, 0)
++
++static const char *wm8980_companding[] = {"Off", "NC", "u-law", "A-law" };
++static const char *wm8980_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
++static const char *wm8980_eqmode[] = {"Capture", "Playback" };
++static const char *wm8980_bw[] = {"Narrow", "Wide" };
++static const char *wm8980_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
++static const char *wm8980_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
++static const char *wm8980_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
++static const char *wm8980_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
++static const char *wm8980_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
++static const char *wm8980_alc[] =
++    {"ALC both on", "ALC left only", "ALC right only", "Limiter" };
++
++static const struct soc_enum wm8980_enum[] = {
++	SOC_ENUM_SINGLE(WM8980_COMP, 1, 4, wm8980_companding), /* adc */
++	SOC_ENUM_SINGLE(WM8980_COMP, 3, 4, wm8980_companding), /* dac */
++	SOC_ENUM_SINGLE(WM8980_DAC,  4, 4, wm8980_deemp),
++	SOC_ENUM_SINGLE(WM8980_EQ1,  8, 2, wm8980_eqmode),
++
++	SOC_ENUM_SINGLE(WM8980_EQ1,  5, 4, wm8980_eq1),
++	SOC_ENUM_SINGLE(WM8980_EQ2,  8, 2, wm8980_bw),
++	SOC_ENUM_SINGLE(WM8980_EQ2,  5, 4, wm8980_eq2),
++	SOC_ENUM_SINGLE(WM8980_EQ3,  8, 2, wm8980_bw),
++
++	SOC_ENUM_SINGLE(WM8980_EQ3,  5, 4, wm8980_eq3),
++	SOC_ENUM_SINGLE(WM8980_EQ4,  8, 2, wm8980_bw),
++	SOC_ENUM_SINGLE(WM8980_EQ4,  5, 4, wm8980_eq4),
++	SOC_ENUM_SINGLE(WM8980_EQ5,  8, 2, wm8980_bw),
++
++	SOC_ENUM_SINGLE(WM8980_EQ5,  5, 4, wm8980_eq5),
++	SOC_ENUM_SINGLE(WM8980_ALC3,  8, 2, wm8980_alc),
++};
++
++static const struct snd_kcontrol_new wm8980_snd_controls[] = {
++SOC_SINGLE("Digital Loopback Switch", WM8980_COMP, 0, 1, 0),
++
++SOC_ENUM("ADC Companding", wm8980_enum[0]),
++SOC_ENUM("DAC Companding", wm8980_enum[1]),
++
++SOC_SINGLE("Jack Detection Enable", WM8980_JACK1, 6, 1, 0),
++
++SOC_SINGLE("DAC Right Inversion Switch", WM8980_DAC, 1, 1, 0),
++SOC_SINGLE("DAC Left Inversion Switch", WM8980_DAC, 0, 1, 0),
++
++SOC_SINGLE("Left Playback Volume", WM8980_DACVOLL, 0, 127, 0),
++SOC_SINGLE("Right Playback Volume", WM8980_DACVOLR, 0, 127, 0),
++
++SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0),
++SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0),
++SOC_SINGLE("High Pass Cut Off", WM8980_ADC, 4, 7, 0),
++SOC_SINGLE("Right ADC Inversion Switch", WM8980_ADC, 1, 1, 0),
++SOC_SINGLE("Left ADC Inversion Switch", WM8980_ADC, 0, 1, 0),
++
++SOC_SINGLE("Left Capture Volume", WM8980_ADCVOLL,  0, 127, 0),
++SOC_SINGLE("Right Capture Volume", WM8980_ADCVOLR,  0, 127, 0),
++
++SOC_ENUM("Equaliser Function", wm8980_enum[3]),
++SOC_ENUM("EQ1 Cut Off", wm8980_enum[4]),
++SOC_SINGLE("EQ1 Volume", WM8980_EQ1,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ2 Bandwith", wm8980_enum[5]),
++SOC_ENUM("EQ2 Cut Off", wm8980_enum[6]),
++SOC_SINGLE("EQ2 Volume", WM8980_EQ2,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ3 Bandwith", wm8980_enum[7]),
++SOC_ENUM("EQ3 Cut Off", wm8980_enum[8]),
++SOC_SINGLE("EQ3 Volume", WM8980_EQ3,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ4 Bandwith", wm8980_enum[9]),
++SOC_ENUM("EQ4 Cut Off", wm8980_enum[10]),
++SOC_SINGLE("EQ4 Volume", WM8980_EQ4,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ5 Bandwith", wm8980_enum[11]),
++SOC_ENUM("EQ5 Cut Off", wm8980_enum[12]),
++SOC_SINGLE("EQ5 Volume", WM8980_EQ5,  0, 31, 1),
++
++SOC_SINGLE("DAC Playback Limiter Switch", WM8980_DACLIM1,  8, 1, 0),
++SOC_SINGLE("DAC Playback Limiter Decay", WM8980_DACLIM1,  4, 15, 0),
++SOC_SINGLE("DAC Playback Limiter Attack", WM8980_DACLIM1,  0, 15, 0),
++
++SOC_SINGLE("DAC Playback Limiter Threshold", WM8980_DACLIM2,  4, 7, 0),
++SOC_SINGLE("DAC Playback Limiter Boost", WM8980_DACLIM2,  0, 15, 0),
++
++SOC_SINGLE("ALC Enable Switch", WM8980_ALC1,  8, 1, 0),
++SOC_SINGLE("ALC Capture Max Gain", WM8980_ALC1,  3, 7, 0),
++SOC_SINGLE("ALC Capture Min Gain", WM8980_ALC1,  0, 7, 0),
++
++SOC_SINGLE("ALC Capture ZC Switch", WM8980_ALC2,  8, 1, 0),
++SOC_SINGLE("ALC Capture Hold", WM8980_ALC2,  4, 7, 0),
++SOC_SINGLE("ALC Capture Target", WM8980_ALC2,  0, 15, 0),
++
++SOC_ENUM("ALC Capture Mode", wm8980_enum[13]),
++SOC_SINGLE("ALC Capture Decay", WM8980_ALC3,  4, 15, 0),
++SOC_SINGLE("ALC Capture Attack", WM8980_ALC3,  0, 15, 0),
++
++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8980_NGATE,  3, 1, 0),
++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8980_NGATE,  0, 7, 0),
++
++SOC_SINGLE("Left Capture PGA ZC Switch", WM8980_INPPGAL,  7, 1, 0),
++SOC_SINGLE("Left Capture PGA Volume", WM8980_INPPGAL,  0, 63, 0),
++
++SOC_SINGLE("Right Capture PGA ZC Switch", WM8980_INPPGAR,  7, 1, 0),
++SOC_SINGLE("Right Capture PGA Volume", WM8980_INPPGAR,  0, 63, 0),
++
++SOC_SINGLE("Left Headphone Playback ZC Switch", WM8980_HPVOLL,  7, 1, 0),
++SOC_SINGLE("Left Headphone Playback Switch", WM8980_HPVOLL,  6, 1, 1),
++SOC_SINGLE("Left Headphone Playback Volume", WM8980_HPVOLL,  0, 63, 0),
++
++SOC_SINGLE("Right Headphone Playback ZC Switch", WM8980_HPVOLR,  7, 1, 0),
++SOC_SINGLE("Right Headphone Playback Switch", WM8980_HPVOLR,  6, 1, 1),
++SOC_SINGLE("Right Headphone Playback Volume", WM8980_HPVOLR,  0, 63, 0),
++
++SOC_SINGLE("Left Speaker Playback ZC Switch", WM8980_SPKVOLL,  7, 1, 0),
++SOC_SINGLE("Left Speaker Playback Switch", WM8980_SPKVOLL,  6, 1, 1),
++SOC_SINGLE("Left Speaker Playback Volume", WM8980_SPKVOLL,  0, 63, 0),
++
++SOC_SINGLE("Right Speaker Playback ZC Switch", WM8980_SPKVOLR,  7, 1, 0),
++SOC_SINGLE("Right Speaker Playback Switch", WM8980_SPKVOLR,  6, 1, 1),
++SOC_SINGLE("Right Speaker Playback Volume", WM8980_SPKVOLR,  0, 63, 0),
++
++SOC_DOUBLE_R("Capture Boost(+20dB)", WM8980_ADCBOOSTL, WM8980_ADCBOOSTR,
++	8, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm8980_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8980_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8980_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* Left Output Mixer */
++static const snd_kcontrol_new_t wm8980_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_OUTPUT, 6, 1, 1),
++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_MIXL, 0, 1, 1),
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXL, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXL, 5, 1, 0),
++};
++
++/* Right Output Mixer */
++static const snd_kcontrol_new_t wm8980_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_OUTPUT, 5, 1, 1),
++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_MIXR, 0, 1, 1),
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXR, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXR, 5, 1, 0),
++};
++
++/* Left AUX Input boost vol */
++static const snd_kcontrol_new_t wm8980_laux_boost_controls =
++SOC_DAPM_SINGLE("Left Aux Volume", WM8980_ADCBOOSTL, 0, 3, 0);
++
++/* Right AUX Input boost vol */
++static const snd_kcontrol_new_t wm8980_raux_boost_controls =
++SOC_DAPM_SINGLE("Right Aux Volume", WM8980_ADCBOOSTR, 0, 3, 0);
++
++/* Left Input boost vol */
++static const snd_kcontrol_new_t wm8980_lmic_boost_controls =
++SOC_DAPM_SINGLE("Left Input Volume", WM8980_ADCBOOSTL, 4, 3, 0);
++
++/* Right Input boost vol */
++static const snd_kcontrol_new_t wm8980_rmic_boost_controls =
++SOC_DAPM_SINGLE("Right Input Volume", WM8980_ADCBOOSTR, 4, 3, 0);
++
++/* Left Aux In to PGA */
++static const snd_kcontrol_new_t wm8980_laux_capture_boost_controls =
++SOC_DAPM_SINGLE("Left Capture Switch", WM8980_ADCBOOSTL,  8, 1, 0);
++
++/* Right  Aux In to PGA */
++static const snd_kcontrol_new_t wm8980_raux_capture_boost_controls =
++SOC_DAPM_SINGLE("Right Capture Switch", WM8980_ADCBOOSTR,  8, 1, 0);
++
++/* Left Input P In to PGA */
++static const snd_kcontrol_new_t wm8980_lmicp_capture_boost_controls =
++SOC_DAPM_SINGLE("Left Input P Capture Boost Switch", WM8980_INPUT,  0, 1, 0);
++
++/* Right Input P In to PGA */
++static const snd_kcontrol_new_t wm8980_rmicp_capture_boost_controls =
++SOC_DAPM_SINGLE("Right Input P Capture Boost Switch", WM8980_INPUT,  4, 1, 0);
++
++/* Left Input N In to PGA */
++static const snd_kcontrol_new_t wm8980_lmicn_capture_boost_controls =
++SOC_DAPM_SINGLE("Left Input N Capture Boost Switch", WM8980_INPUT,  1, 1, 0);
++
++/* Right Input N In to PGA */
++static const snd_kcontrol_new_t wm8980_rmicn_capture_boost_controls =
++SOC_DAPM_SINGLE("Right Input N Capture Boost Switch", WM8980_INPUT,  5, 1, 0);
++
++// TODO Widgets
++static const struct snd_soc_dapm_widget wm8980_dapm_widgets[] = {
++#if 0
++//SND_SOC_DAPM_MUTE("Mono Mute", WM8980_MONOMIX, 6, 0),
++//SND_SOC_DAPM_MUTE("Speaker Mute", WM8980_SPKMIX, 6, 0),
++
++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8980_POWER3, 2, 0,
++	&wm8980_speaker_mixer_controls[0],
++	ARRAY_SIZE(wm8980_speaker_mixer_controls)),
++SND_SOC_DAPM_MIXER("Mono Mixer", WM8980_POWER3, 3, 0,
++	&wm8980_mono_mixer_controls[0],
++	ARRAY_SIZE(wm8980_mono_mixer_controls)),
++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8980_POWER3, 0, 0),
++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8980_POWER3, 0, 0),
++SND_SOC_DAPM_PGA("Aux Input", WM8980_POWER1, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkN Out", WM8980_POWER3, 5, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkP Out", WM8980_POWER3, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out", WM8980_POWER3, 7, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mic PGA", WM8980_POWER2, 2, 0, NULL, 0),
++
++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
++	&wm8980_aux_boost_controls, 1),
++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
++	&wm8980_mic_boost_controls, 1),
++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
++	&wm8980_capture_boost_controls),
++
++SND_SOC_DAPM_MIXER("Boost Mixer", WM8980_POWER2, 4, 0, NULL, 0),
++
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8980_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"),
++#endif
++};
++
++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 wm8980_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(wm8980_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &wm8980_dapm_widgets[i]);
++	}
++
++	/* set up audio path 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;
++}
++
++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},
++	/* TODO: liam - add more entries */
++};
++
++static int wm8980_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	int i;
++	u16 reg;
++
++	if(freq_in == 0 || freq_out == 0) {
++		reg = wm8980_read_reg_cache(codec, WM8980_POWER1);
++		wm8980_write(codec, WM8980_POWER1, reg & 0x1df);
++		return 0;
++	}
++
++	for(i = 0; i < ARRAY_SIZE(pll); i++) {
++		if (freq_in == pll[i].in_hz && freq_out == pll[i].out_hz) {
++			wm8980_write(codec, WM8980_PLLN, (pll[i].pre << 4) | pll[i].n);
++			wm8980_write(codec, WM8980_PLLK1, pll[i].k >> 18);
++			wm8980_write(codec, WM8980_PLLK1, (pll[i].k >> 9) && 0x1ff);
++			wm8980_write(codec, WM8980_PLLK1, pll[i].k && 0x1ff);
++			reg = wm8980_read_reg_cache(codec, WM8980_POWER1);
++			wm8980_write(codec, WM8980_POWER1, reg | 0x020);
++			return 0;
++		}
++	}
++	return -EINVAL;
++}
++
++static int wm8980_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = wm8980_read_reg_cache(codec, WM8980_IFACE) & 0x3;
++	u16 clk = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0xfffe;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		clk |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	wm8980_write(codec, WM8980_IFACE, iface);
++	return 0;
++}
++
++static int wm8980_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 iface = wm8980_read_reg_cache(codec, WM8980_IFACE) & 0xff9f;
++	u16 adn = wm8980_read_reg_cache(codec, WM8980_ADD) & 0x1f1;
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0020;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0040;
++		break;
++	}
++
++	/* filter coefficient */
++	switch (params_rate(params)) {
++	case SNDRV_PCM_RATE_8000:
++		adn |= 0x5 << 1;
++		break;
++	case SNDRV_PCM_RATE_11025:
++		adn |= 0x4 << 1;
++		break;
++	case SNDRV_PCM_RATE_16000:
++		adn |= 0x3 << 1;
++		break;
++	case SNDRV_PCM_RATE_22050:
++		adn |= 0x2 << 1;
++		break;
++	case SNDRV_PCM_RATE_32000:
++		adn |= 0x1 << 1;
++		break;
++	}
++
++	/* set iface */
++	wm8980_write(codec, WM8980_IFACE, iface);
++	wm8980_write(codec, WM8980_ADD, adn);
++	return 0;
++}
++
++static int wm8980_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	switch (div_id) {
++	case WM8980_MCLKDIV:
++		reg = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0x11f;
++		wm8980_write(codec, WM8980_CLOCK, reg | div);
++		break;
++	case WM8980_BCLKDIV:
++		reg = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0x1c7;
++		wm8980_write(codec, WM8980_CLOCK, reg | div);
++		break;
++	case WM8980_OPCLKDIV:
++		reg = wm8980_read_reg_cache(codec, WM8980_GPIO) & 0x1cf;
++		wm8980_write(codec, WM8980_GPIO, reg | div);
++		break;
++	case WM8980_DACOSR:
++		reg = wm8980_read_reg_cache(codec, WM8980_DAC) & 0x1f7;
++		wm8980_write(codec, WM8980_DAC, reg | div);
++		break;
++	case WM8980_ADCOSR:
++		reg = wm8980_read_reg_cache(codec, WM8980_ADC) & 0x1f7;
++		wm8980_write(codec, WM8980_ADC, reg | div);
++		break;
++	case WM8980_MCLKSEL:
++		reg = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0x0ff;
++		wm8980_write(codec, WM8980_CLOCK, reg | div);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static int wm8980_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	u16 mute_reg = wm8980_read_reg_cache(codec, WM8980_DAC) & 0xffbf;
++
++	if(mute)
++		wm8980_write(codec, WM8980_DAC, mute_reg | 0x40);
++	else
++		wm8980_write(codec, WM8980_DAC, mute_reg);
++
++	return 0;
++}
++
++/* TODO: liam need to make this lower power with dapm */
++static int wm8980_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 */
++		wm8980_write(codec, WM8980_POWER1, 0x1ff);
++		wm8980_write(codec, WM8980_POWER2, 0x1ff);
++		wm8980_write(codec, WM8980_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 */
++		wm8980_write(codec, WM8980_POWER1, 0x0);
++		wm8980_write(codec, WM8980_POWER2, 0x0);
++		wm8980_write(codec, WM8980_POWER3, 0x0);
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define WM8980_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 WM8980_FORMATS \
++	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
++	SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE)
++
++struct snd_soc_codec_dai wm8980_dai = {
++	.name = "WM8980 HiFi",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8980_RATES,
++		.formats = WM8980_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8980_RATES,
++		.formats = WM8980_FORMATS,},
++	.ops = {
++		.hw_params = wm8980_hw_params,
++	},
++	.dai_ops = {
++		.digital_mute = wm8980_mute,
++		.set_fmt = wm8980_set_dai_fmt,
++		.set_clkdiv = wm8980_set_dai_clkdiv,
++		.set_pll = wm8980_set_dai_pll,
++	},
++};
++EXPORT_SYMBOL_GPL(wm8980_dai);
++
++static int wm8980_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;
++
++	wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int wm8980_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(wm8980_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8980_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the WM8980 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8980_init(struct snd_soc_device* socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int ret = 0;
++
++	codec->name = "WM8980";
++	codec->owner = THIS_MODULE;
++	codec->read = wm8980_read_reg_cache;
++	codec->write = wm8980_write;
++	codec->dapm_event = wm8980_dapm_event;
++	codec->dai = &wm8980_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(wm8980_reg);
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8980_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, wm8980_reg,
++		sizeof(u16) * ARRAY_SIZE(wm8980_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8980_reg);
++
++	wm8980_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 */
++	wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8980_add_controls(codec);
++	wm8980_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 *wm8980_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8980 2 wire address is 0x1a
++ */
++#define I2C_DRIVERID_WM8980 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 wm8980_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 wm8980_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++	struct snd_soc_device *socdev = wm8980_socdev;
++	struct wm8980_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 = wm8980_init(socdev);
++	if(ret < 0) {
++		err("failed to initialise WM8980\n");
++		goto err;
++	}
++	return ret;
++
++err:
++	kfree(codec);
++	kfree(i2c);
++	return ret;
++
++}
++
++static int wm8980_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 wm8980_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, wm8980_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8980_i2c_driver = {
++	.driver = {
++		.name = "WM8980 I2C Codec",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_WM8980,
++	.attach_adapter = wm8980_i2c_attach,
++	.detach_client =  wm8980_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "WM8980",
++	.driver = &wm8980_i2c_driver,
++};
++#endif
++
++static int wm8980_probe(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct wm8980_setup_data *setup;
++	struct snd_soc_codec *codec;
++	int ret = 0;
++
++	info("WM8980 Audio Codec %s", WM8980_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);
++
++	wm8980_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(&wm8980_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 wm8980_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)
++		wm8980_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(&wm8980_i2c_driver);
++#endif
++	kfree(codec);
++
++	return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8980 = {
++	.probe = 	wm8980_probe,
++	.remove = 	wm8980_remove,
++	.suspend = 	wm8980_suspend,
++	.resume =	wm8980_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8980);
++
++MODULE_DESCRIPTION("ASoC WM8980 driver");
++MODULE_AUTHOR("Mike Arthur");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,116 @@
++/*
++ * wm8980.h  --  WM8980 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 _WM8980_H
++#define _WM8980_H
++
++/* WM8980 register space */
++
++#define WM8980_RESET		0x0
++#define WM8980_POWER1		0x1
++#define WM8980_POWER2		0x2
++#define WM8980_POWER3		0x3
++#define WM8980_IFACE		0x4
++#define WM8980_COMP			0x5
++#define WM8980_CLOCK		0x6
++#define WM8980_ADD			0x7
++#define WM8980_GPIO			0x8
++#define WM8980_JACK1        0x9
++#define WM8980_DAC			0xa
++#define WM8980_DACVOLL	    0xb
++#define WM8980_DACVOLR      0xc
++#define WM8980_JACK2        0xd
++#define WM8980_ADC			0xe
++#define WM8980_ADCVOLL		0xf
++#define WM8980_ADCVOLR      0x10
++#define WM8980_EQ1			0x12
++#define WM8980_EQ2			0x13
++#define WM8980_EQ3			0x14
++#define WM8980_EQ4			0x15
++#define WM8980_EQ5			0x16
++#define WM8980_DACLIM1		0x18
++#define WM8980_DACLIM2		0x19
++#define WM8980_NOTCH1		0x1b
++#define WM8980_NOTCH2		0x1c
++#define WM8980_NOTCH3		0x1d
++#define WM8980_NOTCH4		0x1e
++#define WM8980_ALC1			0x20
++#define WM8980_ALC2			0x21
++#define WM8980_ALC3			0x22
++#define WM8980_NGATE		0x23
++#define WM8980_PLLN			0x24
++#define WM8980_PLLK1		0x25
++#define WM8980_PLLK2		0x26
++#define WM8980_PLLK3		0x27
++#define WM8980_VIDEO		0x28
++#define WM8980_3D           0x29
++#define WM8980_BEEP         0x2b
++#define WM8980_INPUT		0x2c
++#define WM8980_INPPGAL  	0x2d
++#define WM8980_INPPGAR      0x2e
++#define WM8980_ADCBOOSTL	0x2f
++#define WM8980_ADCBOOSTR    0x30
++#define WM8980_OUTPUT		0x31
++#define WM8980_MIXL	        0x32
++#define WM8980_MIXR         0x33
++#define WM8980_HPVOLL		0x34
++#define WM8980_HPVOLR       0x35
++#define WM8980_SPKVOLL      0x36
++#define WM8980_SPKVOLR      0x37
++#define WM8980_OUT3MIX		0x38
++#define WM8980_MONOMIX      0x39
++
++#define WM8980_CACHEREGNUM 	58
++
++/*
++ * WM8980 Clock dividers
++ */
++#define WM8980_MCLKDIV 		0
++#define WM8980_BCLKDIV		1
++#define WM8980_OPCLKDIV		2
++#define WM8980_DACOSR		3
++#define WM8980_ADCOSR		4
++#define WM8980_MCLKSEL		5
++
++#define WM8980_MCLK_MCLK		(0 << 8)
++#define WM8980_MCLK_PLL			(1 << 8)
++
++#define WM8980_MCLK_DIV_1		(0 << 5)
++#define WM8980_MCLK_DIV_1_5		(1 << 5)
++#define WM8980_MCLK_DIV_2		(2 << 5)
++#define WM8980_MCLK_DIV_3		(3 << 5)
++#define WM8980_MCLK_DIV_4		(4 << 5)
++#define WM8980_MCLK_DIV_5_5		(5 << 5)
++#define WM8980_MCLK_DIV_6		(6 << 5)
++
++#define WM8980_BCLK_DIV_1		(0 << 2)
++#define WM8980_BCLK_DIV_2		(1 << 2)
++#define WM8980_BCLK_DIV_4		(2 << 2)
++#define WM8980_BCLK_DIV_8		(3 << 2)
++#define WM8980_BCLK_DIV_16		(4 << 2)
++#define WM8980_BCLK_DIV_32		(5 << 2)
++
++#define WM8980_DACOSR_64		(0 << 3)
++#define WM8980_DACOSR_128		(1 << 3)
++
++#define WM8980_ADCOSR_64		(0 << 3)
++#define WM8980_ADCOSR_128		(1 << 3)
++
++#define WM8980_OPCLK_DIV_1		(0 << 4)
++#define WM8980_OPCLK_DIV_2		(1 << 4)
++#define WM8980_OPCLK_DIV_3		(2 << 4)
++#define WM8980_OPCLK_DIV_4		(3 << 4)
++
++struct wm8980_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8980_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8980;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/eti_b1_wm8731.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/eti_b1_wm8731.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,380 @@
++/*
++ * eti_b1_wm8731  --  SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
++ *
++ * Author:	Frank Mandarino <fmandarino at endrelia.com>
++ *		Endrelia Technologies Inc.
++ * Created:	Mar 29, 2006
++ *
++ * Based on corgi.c by:
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2005 Openedhand Ltd.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *          Richard Purdie <richard at 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/version.h>
++#include <linux/kernel.h>
++#include <linux/clk.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/arch/hardware.h>
++#include <asm/arch/at91_pio.h>
++#include <asm/arch/gpio.h>
++
++#include "../codecs/wm8731.h"
++#include "at91-pcm.h"
++#include "at91-i2s.h"
++
++#if 0
++#define	DBG(x...)	printk(KERN_INFO "eti_b1_wm8731:" x)
++#else
++#define	DBG(x...)
++#endif
++
++#define AT91_PIO_TF1	(1 << (AT91_PIN_PB6 - PIN_BASE) % 32)
++#define AT91_PIO_TK1	(1 << (AT91_PIN_PB7 - PIN_BASE) % 32)
++#define AT91_PIO_TD1	(1 << (AT91_PIN_PB8 - PIN_BASE) % 32)
++#define AT91_PIO_RD1	(1 << (AT91_PIN_PB9 - PIN_BASE) % 32)
++#define AT91_PIO_RK1	(1 << (AT91_PIN_PB10 - PIN_BASE) % 32)
++#define AT91_PIO_RF1	(1 << (AT91_PIN_PB11 - PIN_BASE) % 32)
++
++static struct clk *pck1_clk;
++static struct clk *pllb_clk;
++
++/* current capture and playback sample rates */
++static unsigned int sample_rate[2];
++static spinlock_t sample_rate_lock = SPIN_LOCK_UNLOCKED;
++
++
++static int eti_b1_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	int ret;
++
++	/* cpu clock is the AT91 master clock sent to the SSC */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
++		60000000, SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* codec system clock is supplied by PCK1, set to 12MHz */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK,
++		12000000, SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* Start PCK1 clock. */
++	clk_enable(pck1_clk);
++	DBG("pck1 started\n");
++
++	return 0;
++}
++
++static void eti_b1_shutdown(struct snd_pcm_substream *substream)
++{
++	int dir;
++
++	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
++	sample_rate[dir] = 0;
++
++	/* Stop PCK1 clock. */
++	clk_disable(pck1_clk);
++	DBG("pck1 stopped\n");
++}
++
++static int eti_b1_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	int dir;
++	unsigned int rate, other_rate;
++	int cmr_div, tcmr_period;
++	int ret;
++
++	/*
++	 * The ETI_B1 does not support playback and capture
++	 * at different sample rates because the SSC TF signal supplies
++	 * both ADCLRC and DACLRC.
++	 */
++	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
++	rate = params_rate(params);
++
++	spin_lock_irq(&sample_rate_lock);
++
++	other_rate = sample_rate[1 - dir];
++	if (other_rate > 0 && other_rate != rate) {
++		printk(KERN_WARNING
++			"incompatible substream in other direction\n");
++		spin_unlock_irq(&sample_rate_lock);
++		return -EINVAL;
++	}
++	sample_rate[dir] = rate;
++
++	spin_unlock_irq(&sample_rate_lock);
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/*
++	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
++	 * field divides the system master clock MCK to drive the SSC TK
++	 * signal which provides the codec BCLK.  The TCMR.PERIOD field
++	 * further divides the BCLK signal to drive the SSC TF signal
++	 * which provides the codec DACLRC and ADCLRC signals.
++	 *
++	 * The dividers were determined through trial and error, where a
++	 * CMR.DIV value is chosen such that the resulting BCLK value
++	 * is divisible, or almost divisible, by (2 * sample rate), and
++	 * then TCMR.PERIOD is BCLK / (2 * sample rate) - 1.
++	 *
++	 * The RCMR.PERIOD value could be set to a different value to
++	 * drive the SSC RK pin to allow different DAC and ADC sample
++	 * rates, but on this board the SSC TF signal supplies both LRC
++	 * clocks, so the RCMR.PERIOD is left at the default of 0, which
++	 * disables the divider.
++	 */
++	switch (rate) {
++	case 8000:
++		cmr_div = 25;	   /* BCLK = 60MHz/(2*25) = 1.2MHz */
++		tcmr_period = 74;  /* LRC = BCLK/(2*(74+1)) = 8000Hz */
++		break;
++	case 16000:
++		cmr_div = 7;	   /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
++		tcmr_period = 133; /* LRC = BCLK/(2*(133+1)) = 15991.47Hz */
++		break;
++	case 32000:
++		cmr_div = 7;	   /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
++		tcmr_period = 66;  /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
++		break;
++	case 48000:
++		cmr_div = 13;	   /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
++		tcmr_period = 23;  /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
++		break;
++	default:
++		printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
++		return -EINVAL;
++	}
++
++	/* set the MCK divider for BCLK */
++	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
++	if (ret < 0)
++		return ret;
++
++	/* set the BCLK divider for both DACLRC and ADCLRC */
++	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, tcmr_period);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static struct snd_soc_ops eti_b1_ops = {
++	.startup = eti_b1_startup,
++	.hw_params = eti_b1_hw_params,
++	.shutdown = eti_b1_shutdown,
++};
++
++
++static const struct snd_soc_dapm_widget eti_b1_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 eti_b1_wm8731_init(struct snd_soc_codec *codec)
++{
++	int i;
++
++	DBG("eti_b1_wm8731_init() called\n");
++
++	/* Add specific widgets */
++	for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &eti_b1_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;
++}
++
++static struct snd_soc_dai_link eti_b1_dai = {
++	.name = "WM8731",
++	.stream_name = "WM8731",
++	.cpu_dai = &at91_i2s_dai[1],
++	.codec_dai = &wm8731_dai,
++	.init = eti_b1_wm8731_init,
++	.ops = &eti_b1_ops,
++};
++
++static struct snd_soc_machine snd_soc_machine_eti_b1 = {
++	.name = "ETI_B1",
++	.dai_link = &eti_b1_dai,
++	.num_links = 1,
++};
++
++static struct wm8731_setup_data eti_b1_wm8731_setup = {
++	.i2c_address = 0x1a,
++};
++
++static struct snd_soc_device eti_b1_snd_devdata = {
++	.machine = &snd_soc_machine_eti_b1,
++	.platform = &at91_soc_platform,
++	.codec_dev = &soc_codec_dev_wm8731,
++	.codec_data = &eti_b1_wm8731_setup,
++};
++
++static struct platform_device *eti_b1_snd_device;
++
++static int __init eti_b1_init(void)
++{
++	int ret;
++	u32 ssc_pio_lines;
++	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
++
++	if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
++		DBG("SSC1 memory region is busy\n");
++		return -EBUSY;
++	}
++
++	ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
++	if (!ssc->base) {
++		DBG("SSC1 memory ioremap failed\n");
++		ret = -ENOMEM;
++		goto fail_release_mem;
++	}
++
++	ssc->pid = AT91RM9200_ID_SSC1;
++
++	eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
++	if (!eti_b1_snd_device) {
++		DBG("platform device allocation failed\n");
++		ret = -ENOMEM;
++		goto fail_io_unmap;
++	}
++
++	platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
++	eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
++
++	ret = platform_device_add(eti_b1_snd_device);
++	if (ret) {
++		DBG("platform device add failed\n");
++		platform_device_put(eti_b1_snd_device);
++		goto fail_io_unmap;
++	}
++
++ 	ssc_pio_lines = AT91_PIO_TF1 | AT91_PIO_TK1 | AT91_PIO_TD1
++			| AT91_PIO_RD1 /* | AT91_PIO_RK1 | AT91_PIO_RF1 */;
++
++	/* Reset all PIO registers and assign lines to peripheral A */
++ 	at91_sys_write(AT91_PIOB + PIO_PDR,  ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_ODR,  ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_IDR,  ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_ASR,  ssc_pio_lines);
++ 	at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines);
++
++	/*
++	 * Set PCK1 parent to PLLB and its rate to 12 Mhz.
++	 */
++	pllb_clk = clk_get(NULL, "pllb");
++	pck1_clk = clk_get(NULL, "pck1");
++
++	clk_set_parent(pck1_clk, pllb_clk);
++	clk_set_rate(pck1_clk, 12000000);
++
++	DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
++
++	/* assign the GPIO pin to PCK1 */
++	at91_set_B_periph(AT91_PIN_PA24, 0);
++
++	return ret;
++
++fail_io_unmap:
++	iounmap(ssc->base);
++fail_release_mem:
++	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
++	return ret;
++}
++
++static void __exit eti_b1_exit(void)
++{
++	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
++
++	clk_put(pck1_clk);
++	clk_put(pllb_clk);
++
++	platform_device_unregister(eti_b1_snd_device);
++
++	iounmap(ssc->base);
++	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
++}
++
++module_init(eti_b1_init);
++module_exit(eti_b1_exit);
++
++/* Module information */
++MODULE_AUTHOR("Frank Mandarino <fmandarino at endrelia.com>");
++MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,854 @@
++/*
++ * wm8510.c  --  WM8510 ALSA Soc Audio driver
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ *
++ * Author: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License 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 "wm8510.h"
++
++#define AUDIO_NAME "wm8510"
++#define WM8510_VERSION "0.6"
++
++/*
++ * Debug
++ */
++
++#define WM8510_DEBUG 0
++
++#ifdef WM8510_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_wm8510;
++
++/*
++ * wm8510 register cache
++ * We can't read the WM8510 register space when we are
++ * using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8510_reg[WM8510_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,
++};
++
++/*
++ * read wm8510 register cache
++ */
++static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec * codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg == WM8510_RESET)
++		return 0;
++	if (reg >= WM8510_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write wm8510 register cache
++ */
++static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= WM8510_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the WM8510 register space
++ */
++static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D9 WM8510 register offset
++	 *   D8...D0 register data
++	 */
++	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++	data[1] = value & 0x00ff;
++
++	wm8510_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -EIO;
++}
++
++#define wm8510_reset(c)	wm8510_write(c, WM8510_RESET, 0)
++
++static const char *wm8510_companding[] = {"Off", "NC", "u-law", "A-law" };
++static const char *wm8510_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
++static const char *wm8510_alc[] = {"ALC", "Limiter" };
++
++static const struct soc_enum wm8510_enum[] = {
++	SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */
++	SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */
++	SOC_ENUM_SINGLE(WM8510_DAC,  4, 4, wm8510_deemp),
++	SOC_ENUM_SINGLE(WM8510_ALC3,  8, 2, wm8510_alc),
++};
++
++static const struct snd_kcontrol_new wm8510_snd_controls[] = {
++
++SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0),
++
++SOC_ENUM("DAC Companding", wm8510_enum[1]),
++SOC_ENUM("ADC Companding", wm8510_enum[0]),
++
++SOC_ENUM("Playback De-emphasis", wm8510_enum[2]),
++SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0),
++
++SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0),
++
++SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0),
++SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0),
++SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0),
++
++SOC_SINGLE("Capture Volume", WM8510_ADCVOL,  0, 127, 0),
++
++SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1,  8, 1, 0),
++SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1,  4, 15, 0),
++SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1,  0, 15, 0),
++
++SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2,  4, 7, 0),
++SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2,  0, 15, 0),
++
++SOC_SINGLE("ALC Enable Switch", WM8510_ALC1,  8, 1, 0),
++SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1,  3, 7, 0),
++SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1,  0, 7, 0),
++
++SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2,  8, 1, 0),
++SOC_SINGLE("ALC Capture Hold", WM8510_ALC2,  4, 7, 0),
++SOC_SINGLE("ALC Capture Target", WM8510_ALC2,  0, 15, 0),
++
++SOC_ENUM("ALC Capture Mode", wm8510_enum[3]),
++SOC_SINGLE("ALC Capture Decay", WM8510_ALC3,  4, 15, 0),
++SOC_SINGLE("ALC Capture Attack", WM8510_ALC3,  0, 15, 0),
++
++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE,  3, 1, 0),
++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE,  0, 7, 0),
++
++SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA,  7, 1, 0),
++SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA,  0, 63, 0),
++
++SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL,  7, 1, 0),
++SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL,  6, 1, 1),
++SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL,  0, 63, 0),
++
++SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
++SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm8510_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8510_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* Speaker Output Mixer */
++static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0),
++SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 1),
++};
++
++/* Mono Output Mixer */
++static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = {
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0),
++SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 1),
++};
++
++/* AUX Input boost vol */
++static const struct snd_kcontrol_new wm8510_aux_boost_controls =
++SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0);
++
++/* Mic Input boost vol */
++static const struct snd_kcontrol_new wm8510_mic_boost_controls =
++SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0);
++
++/* Capture boost switch */
++static const struct snd_kcontrol_new wm8510_capture_boost_controls =
++SOC_DAPM_SINGLE("Capture Boost Switch", WM8510_INPPGA,  6, 1, 0);
++
++/* Aux In to PGA */
++static const struct snd_kcontrol_new wm8510_aux_capture_boost_controls =
++SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8510_INPPGA,  2, 1, 0);
++
++/* Mic P In to PGA */
++static const struct snd_kcontrol_new wm8510_micp_capture_boost_controls =
++SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8510_INPPGA,  0, 1, 0);
++
++/* Mic N In to PGA */
++static const struct snd_kcontrol_new wm8510_micn_capture_boost_controls =
++SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8510_INPPGA,  1, 1, 0);
++
++static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = {
++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0,
++	&wm8510_speaker_mixer_controls[0],
++	ARRAY_SIZE(wm8510_speaker_mixer_controls)),
++SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0,
++	&wm8510_mono_mixer_controls[0],
++	ARRAY_SIZE(wm8510_mono_mixer_controls)),
++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0),
++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER3, 0, 0),
++SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, NULL, 0),
++
++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
++	&wm8510_aux_boost_controls, 1),
++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
++	&wm8510_mic_boost_controls, 1),
++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
++	&wm8510_capture_boost_controls),
++
++SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, NULL, 0),
++
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_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 wm8510_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(wm8510_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &wm8510_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 pre_div:4; /* prescale - 1 */
++	unsigned int n:4;
++	unsigned int k;
++};
++
++static struct pll_ pll_div;
++
++/* The size in bits of the pll divide multiplied by 10
++ * to allow rounding later */
++#define FIXED_PLL_SIZE ((1 << 24) * 10)
++
++static void pll_factors(unsigned int target, unsigned int source)
++{
++	unsigned long long Kpart;
++	unsigned int K, Ndiv, Nmod;
++
++	Ndiv = target / source;
++	if (Ndiv < 6) {
++		source >>= 1;
++		pll_div.pre_div = 1;
++		Ndiv = target / source;
++	} else
++		pll_div.pre_div = 0;
++
++	if ((Ndiv < 6) || (Ndiv > 12))
++		printk(KERN_WARNING
++			"WM8510 N value outwith recommended range! N = %d\n",Ndiv);
++
++	pll_div.n = Ndiv;
++	Nmod = target % source;
++	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
++
++	do_div(Kpart, source);
++
++	K = Kpart & 0xFFFFFFFF;
++
++	/* Check if we need to round */
++	if ((K % 10) >= 5)
++		K += 5;
++
++	/* Move down to proper range now rounding is done */
++	K /= 10;
++
++	pll_div.k = K;
++}
++
++static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	if(freq_in == 0 || freq_out == 0) {
++		reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
++		wm8510_write(codec, WM8510_POWER1, reg & 0x1df);
++		return 0;
++	}
++
++	pll_factors(freq_out*8, freq_in);
++
++	wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
++	wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
++	wm8510_write(codec, WM8510_PLLK1, (pll_div.k >> 9) && 0x1ff);
++	wm8510_write(codec, WM8510_PLLK1, pll_div.k && 0x1ff);
++	reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
++	wm8510_write(codec, WM8510_POWER1, reg | 0x020);
++	return 0;
++
++}
++
++/*
++ * Configure WM8510 clock dividers.
++ */
++static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	switch (div_id) {
++	case WM8510_OPCLKDIV:
++		reg = wm8510_read_reg_cache(codec, WM8510_GPIO & 0x1cf);
++		wm8510_write(codec, WM8510_GPIO, reg | div);
++		break;
++	case WM8510_MCLKDIV:
++		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1f);
++		wm8510_write(codec, WM8510_CLOCK, reg | div);
++		break;
++	case WM8510_ADCCLK:
++		reg = wm8510_read_reg_cache(codec, WM8510_ADC & 0x1f7);
++		wm8510_write(codec, WM8510_ADC, reg | div);
++		break;
++	case WM8510_DACCLK:
++		reg = wm8510_read_reg_cache(codec, WM8510_DAC & 0x1f7);
++		wm8510_write(codec, WM8510_DAC, reg | div);
++		break;
++	case WM8510_BCLKDIV:
++		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1e3);
++		wm8510_write(codec, WM8510_CLOCK, reg | div);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = 0;
++	u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		clk |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	wm8510_write(codec, WM8510_IFACE, iface);
++	wm8510_write(codec, WM8510_CLOCK, clk);
++	return 0;
++}
++
++static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 iface = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x19f;
++	u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0020;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0040;
++		break;
++	case SNDRV_PCM_FORMAT_S32_LE:
++		iface |= 0x0060;
++		break;
++	}
++
++	/* filter coefficient */
++	switch (params_rate(params)) {
++	case SNDRV_PCM_RATE_8000:
++		adn |= 0x5 << 1;
++		break;
++	case SNDRV_PCM_RATE_11025:
++		adn |= 0x4 << 1;
++		break;
++	case SNDRV_PCM_RATE_16000:
++		adn |= 0x3 << 1;
++		break;
++	case SNDRV_PCM_RATE_22050:
++		adn |= 0x2 << 1;
++		break;
++	case SNDRV_PCM_RATE_32000:
++		adn |= 0x1 << 1;
++		break;
++	case SNDRV_PCM_RATE_44100:
++		break;
++	}
++
++	wm8510_write(codec, WM8510_IFACE, iface);
++	wm8510_write(codec, WM8510_ADD, adn);
++	return 0;
++}
++
++static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
++
++	if(mute)
++		wm8510_write(codec, WM8510_DAC, mute_reg | 0x40);
++	else
++		wm8510_write(codec, WM8510_DAC, mute_reg);
++	return 0;
++}
++
++/* liam need to make this lower power with dapm */
++static int wm8510_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 */
++		wm8510_write(codec, WM8510_POWER1, 0x1ff);
++		wm8510_write(codec, WM8510_POWER2, 0x1ff);
++		wm8510_write(codec, WM8510_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 */
++		wm8510_write(codec, WM8510_POWER1, 0x0);
++		wm8510_write(codec, WM8510_POWER2, 0x0);
++		wm8510_write(codec, WM8510_POWER3, 0x0);
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define WM8510_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)
++
++#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
++	SNDRV_PCM_FMTBIT_S24_LE)
++
++struct snd_soc_codec_dai wm8510_dai = {
++	.name = "WM8510 HiFi",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8510_RATES,
++		.formats = WM8510_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8510_RATES,
++		.formats = WM8510_FORMATS,},
++	.ops = {
++		.hw_params = wm8510_pcm_hw_params,
++	},
++	.dai_ops = {
++		.digital_mute = wm8510_mute,
++		.set_fmt = wm8510_set_dai_fmt,
++		.set_clkdiv = wm8510_set_dai_clkdiv,
++		.set_pll = wm8510_set_dai_pll,
++	},
++};
++EXPORT_SYMBOL_GPL(wm8510_dai);
++
++static int wm8510_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;
++
++	wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int wm8510_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(wm8510_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8510_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the WM8510 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8510_init(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int ret = 0;
++
++	codec->name = "WM8510";
++	codec->owner = THIS_MODULE;
++	codec->read = wm8510_read_reg_cache;
++	codec->write = wm8510_write;
++	codec->dapm_event = wm8510_dapm_event;
++	codec->dai = &wm8510_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(wm8510_reg);
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8510_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, wm8510_reg,
++		sizeof(u16) * ARRAY_SIZE(wm8510_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8510_reg);
++
++	wm8510_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 */
++	wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8510_add_controls(codec);
++	wm8510_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 *wm8510_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8510 2 wire address is 0x1a
++ */
++#define I2C_DRIVERID_WM8510 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 wm8510_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 wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++	struct snd_soc_device *socdev = wm8510_socdev;
++	struct wm8510_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 = wm8510_init(socdev);
++	if(ret < 0) {
++		err("failed to initialise WM8510\n");
++		goto err;
++	}
++	return ret;
++
++err:
++	kfree(codec);
++	kfree(i2c);
++	return ret;
++}
++
++static int wm8510_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 wm8510_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, wm8510_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8510_i2c_driver = {
++	.driver = {
++		.name = "WM8510 I2C Codec",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_WM8510,
++	.attach_adapter = wm8510_i2c_attach,
++	.detach_client =  wm8510_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "WM8510",
++	.driver = &wm8510_i2c_driver,
++};
++#endif
++
++static int wm8510_probe(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct wm8510_setup_data *setup;
++	struct snd_soc_codec *codec;
++	int ret = 0;
++
++	info("WM8510 Audio Codec %s", WM8510_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);
++
++	wm8510_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(&wm8510_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 wm8510_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)
++		wm8510_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(&wm8510_i2c_driver);
++#endif
++	kfree(codec);
++
++	return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8510 = {
++	.probe = 	wm8510_probe,
++	.remove = 	wm8510_remove,
++	.suspend = 	wm8510_suspend,
++	.resume =	wm8510_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
++
++MODULE_DESCRIPTION("ASoC WM8510 driver");
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,103 @@
++/*
++ * wm8510.h  --  WM8510 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 _WM8510_H
++#define _WM8510_H
++
++/* WM8510 register space */
++
++#define WM8510_RESET		0x0
++#define WM8510_POWER1		0x1
++#define WM8510_POWER2		0x2
++#define WM8510_POWER3		0x3
++#define WM8510_IFACE		0x4
++#define WM8510_COMP			0x5
++#define WM8510_CLOCK		0x6
++#define WM8510_ADD			0x7
++#define WM8510_GPIO			0x8
++#define WM8510_DAC			0xa
++#define WM8510_DACVOL		0xb
++#define WM8510_ADC			0xe
++#define WM8510_ADCVOL		0xf
++#define WM8510_EQ1			0x12
++#define WM8510_EQ2			0x13
++#define WM8510_EQ3			0x14
++#define WM8510_EQ4			0x15
++#define WM8510_EQ5			0x16
++#define WM8510_DACLIM1		0x18
++#define WM8510_DACLIM2		0x19
++#define WM8510_NOTCH1		0x1b
++#define WM8510_NOTCH2		0x1c
++#define WM8510_NOTCH3		0x1d
++#define WM8510_NOTCH4		0x1e
++#define WM8510_ALC1			0x20
++#define WM8510_ALC2			0x21
++#define WM8510_ALC3			0x22
++#define WM8510_NGATE		0x23
++#define WM8510_PLLN			0x24
++#define WM8510_PLLK1		0x25
++#define WM8510_PLLK2		0x26
++#define WM8510_PLLK3		0x27
++#define WM8510_ATTEN		0x28
++#define WM8510_INPUT		0x2c
++#define WM8510_INPPGA		0x2d
++#define WM8510_ADCBOOST		0x2f
++#define WM8510_OUTPUT		0x31
++#define WM8510_SPKMIX		0x32
++#define WM8510_SPKVOL		0x36
++#define WM8510_MONOMIX		0x38
++
++#define WM8510_CACHEREGNUM 	57
++
++/* Clock divider Id's */
++#define WM8510_OPCLKDIV		0
++#define WM8510_MCLKDIV		1
++#define WM8510_ADCCLK		2
++#define WM8510_DACCLK		3
++#define WM8510_BCLKDIV		4
++
++/* DAC clock dividers */
++#define WM8510_DACCLK_F2	(1 << 3)
++#define WM8510_DACCLK_F4	(0 << 3)
++
++/* ADC clock dividers */
++#define WM8510_ADCCLK_F2	(1 << 3)
++#define WM8510_ADCCLK_F4	(0 << 3)
++
++/* PLL Out dividers */
++#define WM8510_OPCLKDIV_1	(0 << 4)
++#define WM8510_OPCLKDIV_2	(1 << 4)
++#define WM8510_OPCLKDIV_3	(2 << 4)
++#define WM8510_OPCLKDIV_4	(3 << 4)
++
++/* BCLK clock dividers */
++#define WM8510_BCLKDIV_1	(0 << 2)
++#define WM8510_BCLKDIV_2	(1 << 2)
++#define WM8510_BCLKDIV_4	(2 << 2)
++#define WM8510_BCLKDIV_8	(3 << 2)
++#define WM8510_BCLKDIV_16	(4 << 2)
++#define WM8510_BCLKDIV_32	(5 << 2)
++
++/* MCLK clock dividers */
++#define WM8510_MCLKDIV_1	(0 << 5)
++#define WM8510_MCLKDIV_1_5	(1 << 5)
++#define WM8510_MCLKDIV_2	(2 << 5)
++#define WM8510_MCLKDIV_3	(3 << 5)
++#define WM8510_MCLKDIV_4	(4 << 5)
++#define WM8510_MCLKDIV_6	(5 << 5)
++#define WM8510_MCLKDIV_8	(6 << 5)
++#define WM8510_MCLKDIV_12	(7 << 5)
++
++struct wm8510_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8510_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8510;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ac97.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ac97.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,222 @@
++/*
++ * imx-ssi.c  --  SSI driver for Freescale IMX
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  Based on mxc-alsa-mc13783 (C) 2006 Freescale.
++ *
++ *  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 Aug 2006   Initial version.
++ *
++ */
++
++
++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = {
++	.name			= "SSI1 PCM Stereo out",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = emi_2_per,
++		.watermark_level = SDMA_TXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI1_STX0,
++		.event_id = DMA_REQ_SSI1_TX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = {
++	.name			= "SSI1 PCM Stereo in",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = per_2_emi,
++		.watermark_level = SDMA_RXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI1_SRX0,
++		.event_id = DMA_REQ_SSI1_RX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = {
++	.name			= "SSI2 PCM Stereo out",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = per_2_emi,
++		.watermark_level = SDMA_TXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI2_STX0,
++		.event_id = DMA_REQ_SSI2_TX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = {
++	.name			= "SSI2 PCM Stereo in",
++	.params = {
++		.bd_number = 1,
++		.transfer_type = per_2_emi,
++		.watermark_level = SDMA_RXFIFO_WATERMARK,
++		.word_size = TRANSFER_16BIT, // maybe add this in setup func
++		.per_address = SSI2_SRX0,
++		.event_id = DMA_REQ_SSI2_RX1,
++		.peripheral_type = SSI,
++	},
++};
++
++static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
++{
++}
++
++static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
++{
++}
++
++static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
++{
++}
++
++static void imx_ssi_ac97_cold_reset(struct snd_ac97 *ac97)
++{
++}
++
++struct snd_ac97_bus_ops soc_ac97_ops = {
++	.read	= imx_ssi_ac97_read,
++	.write	= imx_ssi_ac97_write,
++	.warm_reset	= imx_ssi_ac97_warm_reset,
++	.reset	= imx_ssi_ac97_cold_reset,
++};
++
++
++static intimx_ssi1_ac97_probe(struct platform_device *pdev)
++{
++	int ret;
++
++
++	return ret;
++}
++
++static void imx_ssi1_ac97_remove(struct platform_device *pdev)
++{
++	/* shutdown SSI */
++		if(rtd->cpu_dai->id == 0)
++			SSI1_SCR &= ~SSI_SCR_SSIEN;
++		else
++			SSI2_SCR &= ~SSI_SCR_SSIEN;
++	}
++
++}
++
++static int imx_ssi1_ac97_prepare(struct snd_pcm_substream *substream)
++{
++	// set vra
++}
++
++static int imx_ssi_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++	if (!rtd->cpu_dai->active) {
++
++	}
++
++	return 0;
++}
++
++static int imx_ssi1_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++
++	return ret;
++}
++
++static void imx_ssi_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++
++}
++
++#ifdef CONFIG_PM
++static int imx_ssi_suspend(struct platform_device *dev,
++	struct snd_soc_cpu_dai *dai)
++{
++	if(!dai->active)
++		return 0;
++
++
++	return 0;
++}
++
++static int imx_ssi_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++	if(!dai->active)
++		return 0;
++
++	return 0;
++}
++
++#else
++#define imx_ssi_suspend	NULL
++#define imx_ssi_resume	NULL
++#endif
++
++#define IMX_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)
++
++struct snd_soc_cpu_dai imx_ssi_ac97_dai = {
++	.name = "imx-ac97-1",
++	.id = 0,
++	.type = SND_SOC_DAI_AC97,
++	.suspend = imx_ssi_suspend,
++	.resume = imx_ssi_resume,
++	.playback = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = IMX_AC97_RATES,},
++	.capture = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = IMX_AC97_RATES,},
++	.ops = {
++		.probe = imx_ac97_probe,
++		.remove = imx_ac97_shutdown,
++		.trigger = imx_ssi_trigger,
++		.prepare = imx_ssi_ac97_prepare,},
++},
++{
++	.name = "imx-ac97-2",
++	.id = 1,
++	.type = SND_SOC_DAI_AC97,
++	.suspend = imx_ssi_suspend,
++	.resume = imx_ssi_resume,
++	.playback = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = IMX_AC97_RATES,},
++	.capture = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = IMX_AC97_RATES,},
++	.ops = {
++		.probe = imx_ac97_probe,
++		.remove = imx_ac97_shutdown,
++		.trigger = imx_ssi_trigger,
++		.prepare = imx_ssi_ac97_prepare,},
++};
++
++EXPORT_SYMBOL_GPL(imx_ssi_ac97_dai);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("i.MX ASoC AC97 driver");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/include/linux/i2c-id.h
+===================================================================
+--- linux-2.6.17.14-fic4.test.orig/include/linux/i2c-id.h	2007-01-24 12:19:01.000000000 +0100
++++ linux-2.6.17.14-fic4.test/include/linux/i2c-id.h	2007-01-24 12:21:22.000000000 +0100
+@@ -113,6 +113,9 @@
+ #define I2C_DRIVERID_PCF8563	83	/* Philips PCF8563 RTC		*/
+ #define I2C_DRIVERID_RS5C372	84	/* Ricoh RS5C372 RTC		*/
+ 
++#define I2C_DRIVERID_WM8731	89	/* Wolfson WM8731 audio codec */
++#define I2C_DRIVERID_WM8750	90	/* Wolfson WM8750 audio codec */
++
+ #define I2C_DRIVERID_I2CDEV	900
+ #define I2C_DRIVERID_ARP        902    /* SMBus ARP Client              */
+ #define I2C_DRIVERID_ALERT      903    /* SMBus Alert Responder Client  */
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,882 @@
++/*
++ * wm8976.c  --  WM8976 ALSA Soc Audio driver
++ *
++ * Copyright 2006 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.
++ */
++
++#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 "wm8976.h"
++
++#define AUDIO_NAME "wm8976"
++#define WM8976_VERSION "0.4"
++
++/*
++ * Debug
++ */
++
++#define WM8976_DEBUG 0
++
++#ifdef WM8976_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_wm8976;
++
++/*
++ * wm8976 register cache
++ * We can't read the WM8976 register space when we are
++ * using 2 wire for device control, so we cache them instead.
++ */
++static const u16 wm8976_reg[WM8976_CACHEREGNUM] = {
++    0x0000, 0x0000, 0x0000, 0x0000,
++    0x0050, 0x0000, 0x0140, 0x0000,
++    0x0000, 0x0000, 0x0000, 0x00ff,
++    0x00ff, 0x0000, 0x0100, 0x00ff,
++    0x00ff, 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,
++    0x0033, 0x0010, 0x0010, 0x0100,
++    0x0100, 0x0002, 0x0001, 0x0001,
++    0x0039, 0x0039, 0x0039, 0x0039,
++    0x0001, 0x0001,
++};
++
++/*
++ * read wm8976 register cache
++ */
++static inline unsigned int wm8976_read_reg_cache(struct snd_soc_codec  *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg == WM8976_RESET)
++		return 0;
++	if (reg >= WM8976_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write wm8976 register cache
++ */
++static inline void wm8976_write_reg_cache(struct snd_soc_codec  *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= WM8976_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the WM8976 register space
++ */
++static int wm8976_write(struct snd_soc_codec  *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* data is
++	 *   D15..D9 WM8976 register offset
++	 *   D8...D0 register data
++	 */
++	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
++	data[1] = value & 0x00ff;
++
++	wm8976_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -1;
++}
++
++#define wm8976_reset(c)	wm8976_write(c, WM8976_RESET, 0)
++
++static const char *wm8976_companding[] = {"Off", "NC", "u-law", "A-law" };
++static const char *wm8976_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
++static const char *wm8976_eqmode[] = {"Capture", "Playback" };
++static const char *wm8976_bw[] = {"Narrow", "Wide" };
++static const char *wm8976_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
++static const char *wm8976_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
++static const char *wm8976_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
++static const char *wm8976_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
++static const char *wm8976_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
++static const char *wm8976_alc[] =
++    {"ALC both on", "ALC left only", "ALC right only", "Limiter" };
++
++static const struct soc_enum wm8976_enum[] = {
++	SOC_ENUM_SINGLE(WM8976_COMP, 1, 4, wm8976_companding), /* adc */
++	SOC_ENUM_SINGLE(WM8976_COMP, 3, 4, wm8976_companding), /* dac */
++	SOC_ENUM_SINGLE(WM8976_DAC,  4, 4, wm8976_deemp),
++	SOC_ENUM_SINGLE(WM8976_EQ1,  8, 2, wm8976_eqmode),
++
++	SOC_ENUM_SINGLE(WM8976_EQ1,  5, 4, wm8976_eq1),
++	SOC_ENUM_SINGLE(WM8976_EQ2,  8, 2, wm8976_bw),
++	SOC_ENUM_SINGLE(WM8976_EQ2,  5, 4, wm8976_eq2),
++	SOC_ENUM_SINGLE(WM8976_EQ3,  8, 2, wm8976_bw),
++
++	SOC_ENUM_SINGLE(WM8976_EQ3,  5, 4, wm8976_eq3),
++	SOC_ENUM_SINGLE(WM8976_EQ4,  8, 2, wm8976_bw),
++	SOC_ENUM_SINGLE(WM8976_EQ4,  5, 4, wm8976_eq4),
++	SOC_ENUM_SINGLE(WM8976_EQ5,  8, 2, wm8976_bw),
++
++	SOC_ENUM_SINGLE(WM8976_EQ5,  5, 4, wm8976_eq5),
++	SOC_ENUM_SINGLE(WM8976_ALC3,  8, 2, wm8976_alc),
++};
++
++static const struct snd_kcontrol_new wm8976_snd_controls[] = {
++SOC_SINGLE("Digital Loopback Switch", WM8976_COMP, 0, 1, 0),
++
++SOC_ENUM("ADC Companding", wm8976_enum[0]),
++SOC_ENUM("DAC Companding", wm8976_enum[1]),
++
++SOC_SINGLE("Jack Detection Enable", WM8976_JACK1, 6, 1, 0),
++
++SOC_DOUBLE("DAC Inversion Switch", WM8976_DAC, 0, 1, 1, 0),
++
++SOC_DOUBLE_R("Headphone Playback Volume", WM8976_DACVOLL, WM8976_DACVOLR, 0, 127, 0),
++
++SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0),
++SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0),
++SOC_SINGLE("High Pass Cut Off", WM8976_ADC, 4, 7, 0),
++
++SOC_DOUBLE("ADC Inversion Switch", WM8976_ADC, 0, 1, 1, 0),
++
++SOC_SINGLE("Capture Volume", WM8976_ADCVOL,  0, 127, 0),
++
++SOC_ENUM("Equaliser Function", wm8976_enum[3]),
++SOC_ENUM("EQ1 Cut Off", wm8976_enum[4]),
++SOC_SINGLE("EQ1 Volume", WM8976_EQ1,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ2 Bandwith", wm8976_enum[5]),
++SOC_ENUM("EQ2 Cut Off", wm8976_enum[6]),
++SOC_SINGLE("EQ2 Volume", WM8976_EQ2,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ3 Bandwith", wm8976_enum[7]),
++SOC_ENUM("EQ3 Cut Off", wm8976_enum[8]),
++SOC_SINGLE("EQ3 Volume", WM8976_EQ3,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ4 Bandwith", wm8976_enum[9]),
++SOC_ENUM("EQ4 Cut Off", wm8976_enum[10]),
++SOC_SINGLE("EQ4 Volume", WM8976_EQ4,  0, 31, 1),
++
++SOC_ENUM("Equaliser EQ5 Bandwith", wm8976_enum[11]),
++SOC_ENUM("EQ5 Cut Off", wm8976_enum[12]),
++SOC_SINGLE("EQ5 Volume", WM8976_EQ5,  0, 31, 1),
++
++SOC_SINGLE("DAC Playback Limiter Switch", WM8976_DACLIM1,  8, 1, 0),
++SOC_SINGLE("DAC Playback Limiter Decay", WM8976_DACLIM1,  4, 15, 0),
++SOC_SINGLE("DAC Playback Limiter Attack", WM8976_DACLIM1,  0, 15, 0),
++
++SOC_SINGLE("DAC Playback Limiter Threshold", WM8976_DACLIM2,  4, 7, 0),
++SOC_SINGLE("DAC Playback Limiter Boost", WM8976_DACLIM2,  0, 15, 0),
++
++SOC_SINGLE("ALC Enable Switch", WM8976_ALC1,  8, 1, 0),
++SOC_SINGLE("ALC Capture Max Gain", WM8976_ALC1,  3, 7, 0),
++SOC_SINGLE("ALC Capture Min Gain", WM8976_ALC1,  0, 7, 0),
++
++SOC_SINGLE("ALC Capture ZC Switch", WM8976_ALC2,  8, 1, 0),
++SOC_SINGLE("ALC Capture Hold", WM8976_ALC2,  4, 7, 0),
++SOC_SINGLE("ALC Capture Target", WM8976_ALC2,  0, 15, 0),
++
++SOC_ENUM("ALC Capture Mode", wm8976_enum[13]),
++SOC_SINGLE("ALC Capture Decay", WM8976_ALC3,  4, 15, 0),
++SOC_SINGLE("ALC Capture Attack", WM8976_ALC3,  0, 15, 0),
++
++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8976_NGATE,  3, 1, 0),
++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8976_NGATE,  0, 7, 0),
++
++SOC_SINGLE("Capture PGA ZC Switch", WM8976_INPPGA,  7, 1, 0),
++SOC_SINGLE("Capture PGA Volume", WM8976_INPPGA,  0, 63, 0),
++
++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8976_HPVOLL,  WM8976_HPVOLR, 7, 1, 0),
++SOC_DOUBLE_R("Headphone Playback Switch", WM8976_HPVOLL,  WM8976_HPVOLR, 6, 1, 1),
++SOC_DOUBLE_R("Headphone Playback Volume", WM8976_HPVOLL,  WM8976_HPVOLR, 0, 63, 0),
++
++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8976_SPKVOLL,  WM8976_SPKVOLR, 7, 1, 0),
++SOC_DOUBLE_R("Speaker Playback Switch", WM8976_SPKVOLL,  WM8976_SPKVOLR, 6, 1, 1),
++SOC_DOUBLE_R("Speaker Playback Volume", WM8976_SPKVOLL,  WM8976_SPKVOLR, 0, 63, 0),
++
++SOC_SINGLE("Capture Boost(+20dB)", WM8976_ADCBOOST, 8, 1, 0),
++};
++
++/* add non dapm controls */
++static int wm8976_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(wm8976_snd_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8976_snd_controls[i],codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* Left Output Mixer */
++static const snd_kcontrol_new_t wm8976_left_mixer_controls[] = {
++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_OUTPUT, 6, 1, 1),
++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_MIXL, 0, 1, 1),
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXL, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXL, 5, 1, 0),
++};
++
++/* Right Output Mixer */
++static const snd_kcontrol_new_t wm8976_right_mixer_controls[] = {
++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_OUTPUT, 5, 1, 1),
++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_MIXR, 0, 1, 1),
++SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXR, 1, 1, 0),
++SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXR, 5, 1, 0),
++};
++
++/* Left AUX Input boost vol */
++static const snd_kcontrol_new_t wm8976_laux_boost_controls =
++SOC_DAPM_SINGLE("Aux Volume", WM8976_ADCBOOST, 0, 3, 0);
++
++/* Left Input boost vol */
++static const snd_kcontrol_new_t wm8976_lmic_boost_controls =
++SOC_DAPM_SINGLE("Input Volume", WM8976_ADCBOOST, 4, 3, 0);
++
++/* Left Aux In to PGA */
++static const snd_kcontrol_new_t wm8976_laux_capture_boost_controls =
++SOC_DAPM_SINGLE("Capture Switch", WM8976_ADCBOOST,  8, 1, 0);
++
++/* Left Input P In to PGA */
++static const snd_kcontrol_new_t wm8976_lmicp_capture_boost_controls =
++SOC_DAPM_SINGLE("Input P Capture Boost Switch", WM8976_INPUT,  0, 1, 0);
++
++/* Left Input N In to PGA */
++static const snd_kcontrol_new_t wm8976_lmicn_capture_boost_controls =
++SOC_DAPM_SINGLE("Input N Capture Boost Switch", WM8976_INPUT,  1, 1, 0);
++
++// TODO Widgets
++static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = {
++#if 0
++//SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0),
++//SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0),
++
++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0,
++	&wm8976_speaker_mixer_controls[0],
++	ARRAY_SIZE(wm8976_speaker_mixer_controls)),
++SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0,
++	&wm8976_mono_mixer_controls[0],
++	ARRAY_SIZE(wm8976_mono_mixer_controls)),
++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0),
++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0),
++SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0),
++SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0),
++SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0),
++
++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
++	&wm8976_aux_boost_controls, 1),
++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
++	&wm8976_mic_boost_controls, 1),
++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
++	&wm8976_capture_boost_controls),
++
++SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0),
++
++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_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"),
++#endif
++};
++
++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 wm8976_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]);
++	}
++
++	/* set up audio path 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;
++}
++
++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},
++	/* TODO: liam - add more entries */
++};
++
++static int wm8976_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
++		int pll_id, unsigned int freq_in, unsigned int freq_out)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	int i;
++	u16 reg;
++
++	if(freq_in == 0 || freq_out == 0) {
++		reg = wm8976_read_reg_cache(codec, WM8976_POWER1);
++		wm8976_write(codec, WM8976_POWER1, reg & 0x1df);
++		return 0;
++	}
++
++	for(i = 0; i < ARRAY_SIZE(pll); i++) {
++		if (freq_in == pll[i].in_hz && freq_out == pll[i].out_hz) {
++			wm8976_write(codec, WM8976_PLLN, (pll[i].pre << 4) | pll[i].n);
++			wm8976_write(codec, WM8976_PLLK1, pll[i].k >> 18);
++			wm8976_write(codec, WM8976_PLLK1, (pll[i].k >> 9) && 0x1ff);
++			wm8976_write(codec, WM8976_PLLK1, pll[i].k && 0x1ff);
++			reg = wm8976_read_reg_cache(codec, WM8976_POWER1);
++			wm8976_write(codec, WM8976_POWER1, reg | 0x020);
++			return 0;
++		}
++	}
++	return -EINVAL;
++}
++
++static int wm8976_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++		unsigned int fmt)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 iface = wm8976_read_reg_cache(codec, WM8976_IFACE) & 0x3;
++	u16 clk = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0xfffe;
++
++	/* set master/slave audio interface */
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		clk |= 0x0001;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	/* interface format */
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		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;
++	default:
++		return -EINVAL;
++	}
++
++	/* clock inversion */
++	switch (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;
++	default:
++		return -EINVAL;
++	}
++
++	wm8976_write(codec, WM8976_IFACE, iface);
++	return 0;
++}
++
++static int wm8976_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_device *socdev = rtd->socdev;
++	struct snd_soc_codec *codec = socdev->codec;
++	u16 iface = wm8976_read_reg_cache(codec, WM8976_IFACE) & 0xff9f;
++	u16 adn = wm8976_read_reg_cache(codec, WM8976_ADD) & 0x1f1;
++
++	/* bit size */
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S16_LE:
++		break;
++	case SNDRV_PCM_FORMAT_S20_3LE:
++		iface |= 0x0020;
++		break;
++	case SNDRV_PCM_FORMAT_S24_LE:
++		iface |= 0x0040;
++		break;
++	}
++
++	/* filter coefficient */
++	switch (params_rate(params)) {
++	case SNDRV_PCM_RATE_8000:
++		adn |= 0x5 << 1;
++		break;
++	case SNDRV_PCM_RATE_11025:
++		adn |= 0x4 << 1;
++		break;
++	case SNDRV_PCM_RATE_16000:
++		adn |= 0x3 << 1;
++		break;
++	case SNDRV_PCM_RATE_22050:
++		adn |= 0x2 << 1;
++		break;
++	case SNDRV_PCM_RATE_32000:
++		adn |= 0x1 << 1;
++		break;
++	}
++
++	/* set iface */
++	wm8976_write(codec, WM8976_IFACE, iface);
++	wm8976_write(codec, WM8976_ADD, adn);
++	return 0;
++}
++
++static int wm8976_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
++		int div_id, int div)
++{
++	struct snd_soc_codec *codec = codec_dai->codec;
++	u16 reg;
++
++	switch (div_id) {
++	case WM8976_MCLKDIV:
++		reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x11f;
++		wm8976_write(codec, WM8976_CLOCK, reg | div);
++		break;
++	case WM8976_BCLKDIV:
++		reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x1c7;
++		wm8976_write(codec, WM8976_CLOCK, reg | div);
++		break;
++	case WM8976_OPCLKDIV:
++		reg = wm8976_read_reg_cache(codec, WM8976_GPIO) & 0x1cf;
++		wm8976_write(codec, WM8976_GPIO, reg | div);
++		break;
++	case WM8976_DACOSR:
++		reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0x1f7;
++		wm8976_write(codec, WM8976_DAC, reg | div);
++		break;
++	case WM8976_ADCOSR:
++		reg = wm8976_read_reg_cache(codec, WM8976_ADC) & 0x1f7;
++		wm8976_write(codec, WM8976_ADC, reg | div);
++		break;
++	case WM8976_MCLKSEL:
++		reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x0ff;
++		wm8976_write(codec, WM8976_CLOCK, reg | div);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static int wm8976_mute(struct snd_soc_codec_dai *dai, int mute)
++{
++	struct snd_soc_codec *codec = dai->codec;
++	u16 mute_reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0xffbf;
++
++	if(mute)
++		wm8976_write(codec, WM8976_DAC, mute_reg | 0x40);
++	else
++		wm8976_write(codec, WM8976_DAC, mute_reg);
++
++	return 0;
++}
++
++/* TODO: liam need to make this lower power with dapm */
++static int wm8976_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 */
++		wm8976_write(codec, WM8976_POWER1, 0x1ff);
++		wm8976_write(codec, WM8976_POWER2, 0x1ff);
++		wm8976_write(codec, WM8976_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 */
++		wm8976_write(codec, WM8976_POWER1, 0x0);
++		wm8976_write(codec, WM8976_POWER2, 0x0);
++		wm8976_write(codec, WM8976_POWER3, 0x0);
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++#define WM8976_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 WM8976_FORMATS \
++	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
++	SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE)
++
++struct snd_soc_codec_dai wm8976_dai = {
++	.name = "WM8976 HiFi",
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++		.rates = WM8976_RATES,
++		.formats = WM8976_FORMATS,},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = WM8976_RATES,
++		.formats = WM8976_FORMATS,},
++	.ops = {
++		.hw_params = wm8976_hw_params,
++	},
++	.dai_ops = {
++		.digital_mute = wm8976_mute,
++		.set_fmt = wm8976_set_dai_fmt,
++		.set_clkdiv = wm8976_set_dai_clkdiv,
++		.set_pll = wm8976_set_dai_pll,
++	},
++};
++EXPORT_SYMBOL_GPL(wm8976_dai);
++
++static int wm8976_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;
++
++	wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int wm8976_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(wm8976_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8976_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the WM8976 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int wm8976_init(struct snd_soc_device* socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int ret = 0;
++
++	codec->name = "WM8976";
++	codec->owner = THIS_MODULE;
++	codec->read = wm8976_read_reg_cache;
++	codec->write = wm8976_write;
++	codec->dapm_event = wm8976_dapm_event;
++	codec->dai = &wm8976_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(wm8976_reg);
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8976_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache, wm8976_reg,
++		sizeof(u16) * ARRAY_SIZE(wm8976_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8976_reg);
++
++	wm8976_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 */
++	wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	wm8976_add_controls(codec);
++	wm8976_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 *wm8976_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * WM8976 2 wire address is 0x1a
++ */
++#define I2C_DRIVERID_WM8976 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 wm8976_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 wm8976_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++	struct snd_soc_device *socdev = wm8976_socdev;
++	struct wm8976_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 = wm8976_init(socdev);
++	if(ret < 0) {
++		err("failed to initialise WM8976\n");
++		goto err;
++	}
++	return ret;
++
++err:
++	kfree(codec);
++	kfree(i2c);
++	return ret;
++
++}
++
++static int wm8976_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 wm8976_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, wm8976_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver wm8976_i2c_driver = {
++	.driver = {
++		.name = "WM8976 I2C Codec",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_WM8976,
++	.attach_adapter = wm8976_i2c_attach,
++	.detach_client =  wm8976_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "WM8976",
++	.driver = &wm8976_i2c_driver,
++};
++#endif
++
++static int wm8976_probe(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct wm8976_setup_data *setup;
++	struct snd_soc_codec *codec;
++	int ret = 0;
++
++	info("WM8976 Audio Codec %s", WM8976_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);
++
++	wm8976_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(&wm8976_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 wm8976_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)
++		wm8976_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(&wm8976_i2c_driver);
++#endif
++	kfree(codec);
++
++	return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_wm8976 = {
++	.probe = 	wm8976_probe,
++	.remove = 	wm8976_remove,
++	.suspend = 	wm8976_suspend,
++	.resume =	wm8976_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8976);
++
++MODULE_DESCRIPTION("ASoC WM8976 driver");
++MODULE_AUTHOR("Graeme Gregory");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,112 @@
++/*
++ * wm8976.h  --  WM8976 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 _WM8976_H
++#define _WM8976_H
++
++/* WM8976 register space */
++
++#define WM8976_RESET		0x0
++#define WM8976_POWER1		0x1
++#define WM8976_POWER2		0x2
++#define WM8976_POWER3		0x3
++#define WM8976_IFACE		0x4
++#define WM8976_COMP			0x5
++#define WM8976_CLOCK		0x6
++#define WM8976_ADD			0x7
++#define WM8976_GPIO			0x8
++#define WM8976_JACK1        0x9
++#define WM8976_DAC			0xa
++#define WM8976_DACVOLL	    0xb
++#define WM8976_DACVOLR      0xc
++#define WM8976_JACK2        0xd
++#define WM8976_ADC			0xe
++#define WM8976_ADCVOL		0xf
++#define WM8976_EQ1			0x12
++#define WM8976_EQ2			0x13
++#define WM8976_EQ3			0x14
++#define WM8976_EQ4			0x15
++#define WM8976_EQ5			0x16
++#define WM8976_DACLIM1		0x18
++#define WM8976_DACLIM2		0x19
++#define WM8976_NOTCH1		0x1b
++#define WM8976_NOTCH2		0x1c
++#define WM8976_NOTCH3		0x1d
++#define WM8976_NOTCH4		0x1e
++#define WM8976_ALC1			0x20
++#define WM8976_ALC2			0x21
++#define WM8976_ALC3			0x22
++#define WM8976_NGATE		0x23
++#define WM8976_PLLN			0x24
++#define WM8976_PLLK1		0x25
++#define WM8976_PLLK2		0x26
++#define WM8976_PLLK3		0x27
++#define WM8976_3D           0x29
++#define WM8976_BEEP         0x2b
++#define WM8976_INPUT		0x2c
++#define WM8976_INPPGA	  	0x2d
++#define WM8976_ADCBOOST		0x2f
++#define WM8976_OUTPUT		0x31
++#define WM8976_MIXL	        0x32
++#define WM8976_MIXR         0x33
++#define WM8976_HPVOLL		0x34
++#define WM8976_HPVOLR       0x35
++#define WM8976_SPKVOLL      0x36
++#define WM8976_SPKVOLR      0x37
++#define WM8976_OUT3MIX		0x38
++#define WM8976_MONOMIX      0x39
++
++#define WM8976_CACHEREGNUM 	58
++
++/*
++ * WM8976 Clock dividers
++ */
++#define WM8976_MCLKDIV 		0
++#define WM8976_BCLKDIV		1
++#define WM8976_OPCLKDIV		2
++#define WM8976_DACOSR		3
++#define WM8976_ADCOSR		4
++#define WM8976_MCLKSEL		5
++
++#define WM8976_MCLK_MCLK		(0 << 8)
++#define WM8976_MCLK_PLL			(1 << 8)
++
++#define WM8976_MCLK_DIV_1		(0 << 5)
++#define WM8976_MCLK_DIV_1_5		(1 << 5)
++#define WM8976_MCLK_DIV_2		(2 << 5)
++#define WM8976_MCLK_DIV_3		(3 << 5)
++#define WM8976_MCLK_DIV_4		(4 << 5)
++#define WM8976_MCLK_DIV_5_5		(5 << 5)
++#define WM8976_MCLK_DIV_6		(6 << 5)
++
++#define WM8976_BCLK_DIV_1		(0 << 2)
++#define WM8976_BCLK_DIV_2		(1 << 2)
++#define WM8976_BCLK_DIV_4		(2 << 2)
++#define WM8976_BCLK_DIV_8		(3 << 2)
++#define WM8976_BCLK_DIV_16		(4 << 2)
++#define WM8976_BCLK_DIV_32		(5 << 2)
++
++#define WM8976_DACOSR_64		(0 << 3)
++#define WM8976_DACOSR_128		(1 << 3)
++
++#define WM8976_ADCOSR_64		(0 << 3)
++#define WM8976_ADCOSR_128		(1 << 3)
++
++#define WM8976_OPCLK_DIV_1		(0 << 4)
++#define WM8976_OPCLK_DIV_2		(1 << 4)
++#define WM8976_OPCLK_DIV_3		(2 << 4)
++#define WM8976_OPCLK_DIV_4		(3 << 4)
++
++struct wm8976_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai wm8976_dai;
++extern struct snd_soc_codec_device soc_codec_dev_wm8976;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,454 @@
++/*
++ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ * Based on pxa2xx-pcm.c by	Nicolas Pitre, (C) 2004 MontaVista Software, Inc.
++ * and on mxc-alsa-mc13783 (C) 2006 Freescale.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ *  Revision history
++ *    29th Aug 2006   Initial version.
++ *
++ */
++
++#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 "imx-pcm.h"
++
++/* debug */
++#define IMX_DEBUG 0
++#if IMX_DEBUG
++#define dbg(format, arg...) printk(format, ## arg)
++#else
++#define dbg(format, arg...)
++#endif
++
++static const struct snd_pcm_hardware mxc_pcm_hardware = {
++	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
++				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
++				   SNDRV_PCM_INFO_MMAP |
++				   SNDRV_PCM_INFO_MMAP_VALID |
++				   SNDRV_PCM_INFO_PAUSE |
++				   SNDRV_PCM_INFO_RESUME),
++	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
++					SNDRV_PCM_FMTBIT_S24_LE,
++	.buffer_bytes_max	= 32 * 1024,
++	.period_bytes_min	= 64,
++	.period_bytes_max	= 8 * 1024,
++	.periods_min		= 2,
++	.periods_max		= 255,
++	.fifo_size		= 0,
++};
++
++struct mxc_runtime_data {
++	int dma_ch;
++	struct mxc_pcm_dma_param *dma_params;
++};
++
++/*!
++  * This function stops the current dma transfert for playback
++  * and clears the dma pointers.
++  *
++  * @param	substream	pointer to the structure of the current stream.
++  *
++  */
++static void audio_stop_dma(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size);
++	unsigned int offset  dma_size * s->periods;
++	unsigned long flags;
++
++	spin_lock_irqsave(&prtd->dma_lock, flags);
++
++	dbg("MXC : audio_stop_dma active = 0\n");
++	prtd->active = 0;
++	prtd->period = 0;
++	prtd->periods = 0;
++
++	/* this stops the dma channel and clears the buffer ptrs */
++	mxc_dma_stop(prtd->dma_wchannel);
++	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_TO_DEVICE);
++	else
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_FROM_DEVICE);
++
++	spin_unlock_irqrestore(&prtd->dma_lock, flags);
++}
++
++/*!
++  * This function is called whenever a new audio block needs to be
++  * transferred to mc13783. The function receives the address and the size
++  * of the new block and start a new DMA transfer.
++  *
++  * @param	substream	pointer to the structure of the current stream.
++  *
++  */
++static int dma_new_period(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime =  substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	unsigned int dma_size;
++	unsigned int offset;
++	int ret=0;
++	dma_request_t sdma_request;
++
++	if (prtd->active){
++		memset(&sdma_request, 0, sizeof(dma_request_t));
++		dma_size = frames_to_bytes(runtime, runtime->period_size);
++	    dbg("s->period (%x) runtime->periods (%d)\n",
++			s->period,runtime->periods);
++		dbg("runtime->period_size (%d) dma_size (%d)\n",
++			(unsigned int)runtime->period_size,
++			runtime->dma_bytes);
++
++       	offset = dma_size * prtd->period;
++		snd_assert(dma_size <= DMA_BUF_SIZE, );
++		if(substream == SNDRV_PCM_STREAM_PLAYBACK)
++			sdma_request.sourceAddr = (char*)(dma_map_single(NULL,
++				runtime->dma_area + offset, dma_size, DMA_TO_DEVICE));
++		else
++			sdma_request.destAddr = (char*)(dma_map_single(NULL,
++				runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE));
++		sdma_request.count = dma_size;
++
++		dbg("MXC: Start DMA offset (%d) size (%d)\n", offset,
++						 runtime->dma_bytes);
++
++       	mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0);
++		if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) {
++			dbg("audio_process_dma: cannot queue DMA buffer\
++							(%i)\n", ret);
++			return err;
++		}
++		prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */
++		prtd->period++;
++		prtd->period %= runtime->periods;
++    }
++	return ret;
++}
++
++
++/*!
++  * This is a callback which will be called
++  * when a TX transfer finishes. The call occurs
++  * in interrupt context.
++  *
++  * @param	dat	pointer to the structure of the current stream.
++  *
++  */
++static void audio_dma_irq(void *data)
++{
++	struct snd_pcm_substream *substream;
++	struct snd_pcm_runtime *runtime;
++	struct mxc_runtime_data *prtd;
++	unsigned int dma_size;
++	unsigned int previous_period;
++	unsigned int offset;
++
++	substream = data;
++	runtime = substream->runtime;
++	prtd = runtime->private_data;
++	previous_period  = prtd->periods;
++	dma_size = frames_to_bytes(runtime, runtime->period_size);
++	offset = dma_size * previous_period;
++
++	prtd->tx_spin = 0;
++	prtd->periods++;
++	prtd->periods %= runtime->periods;
++
++	/*
++	  * Give back to the CPU the access to the non cached memory
++	  */
++	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_TO_DEVICE);
++	else
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_FROM_DEVICE);
++	/*
++	  * If we are getting a callback for an active stream then we inform
++	  * the PCM middle layer we've finished a period
++	  */
++ 	if (prtd->active)
++		snd_pcm_period_elapsed(substream);
++
++	/*
++	  * Trig next DMA transfer
++	  */
++	dma_new_period(substream);
++}
++
++/*!
++  * This function configures the hardware to allow audio
++  * playback operations. It is called by ALSA framework.
++  *
++  * @param	substream	pointer to the structure of the current stream.
++  *
++  * @return              0 on success, -1 otherwise.
++  */
++static int
++snd_mxc_prepare(struct snd_pcm_substream *substream)
++{
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	int ret = 0;
++	prtd->period = 0;
++	prtd->periods = 0;
++
++	dma_channel_params params;
++	int channel = 0; // passed in ?
++
++	if ((ret  = mxc_request_dma(&channel, "ALSA TX SDMA") < 0)){
++		dbg("error requesting a write dma channel\n");
++		return ret;
++	}
++
++	/* configure DMA params */
++	memset(&params, 0, sizeof(dma_channel_params));
++	params.bd_number = 1;
++	params.arg = s;
++	params.callback = callback;
++	params.transfer_type = emi_2_per;
++	params.watermark_level = SDMA_TXFIFO_WATERMARK;
++	params.word_size = TRANSFER_16BIT;
++	//dbg(KERN_ERR "activating connection SSI1 - SDMA\n");
++	params.per_address = SSI1_BASE_ADDR;
++	params.event_id = DMA_REQ_SSI1_TX1;
++	params.peripheral_type = SSI;
++
++	/* set up chn with params */
++	mxc_dma_setup_channel(channel, &params);
++	s->dma_wchannel = channel;
++
++	return ret;
++}
++
++static int mxc_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	int ret;
++
++	if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
++		return ret;
++	runtime->dma_addr = virt_to_phys(runtime->dma_area);
++
++	return ret;
++}
++
++static int mxc_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++	return snd_pcm_lib_free_pages(substream);
++}
++
++static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct mxc_runtime_data *prtd = substream->runtime->private_data;
++	int ret = 0;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		prtd->tx_spin = 0;
++		/* requested stream startup */
++		prtd->active = 1;
++        ret = dma_new_period(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		/* requested stream shutdown */
++		ret = audio_stop_dma(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++		prtd->active = 0;
++		prtd->periods = 0;
++		break;
++	case SNDRV_PCM_TRIGGER_RESUME:
++		prtd->active = 1;
++		prtd->tx_spin = 0;
++		ret = dma_new_period(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		prtd->active = 0;
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		prtd->active = 1;
++		if (prtd->old_offset) {
++			prtd->tx_spin = 0;
++            ret = dma_new_period(substream);
++		}
++		break;
++	default:
++		ret = -EINVAL;
++		break;
++	}
++
++	return ret;
++}
++
++static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	unsigned int offset = 0;
++
++	/* tx_spin value is used here to check if a transfert is active */
++	if (prtd->tx_spin){
++		offset = (runtime->period_size * (prtd->periods)) +
++						(runtime->period_size >> 1);
++		if (offset >= runtime->buffer_size)
++			offset = runtime->period_size >> 1;
++	} else {
++		offset = (runtime->period_size * (s->periods));
++		if (offset >= runtime->buffer_size)
++			offset = 0;
++	}
++
++	return offset;
++}
++
++
++static int mxc_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd;
++	int ret;
++
++	snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware);
++
++	if ((err = snd_pcm_hw_constraint_integer(runtime,
++					SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
++		return err;
++	if ((err = snd_pcm_hw_constraint_list(runtime, 0,
++			SNDRV_PCM_HW_PARAM_RATE, &hw_playback_rates)) < 0)
++		return err;
++	msleep(10); // liam - why
++
++	/* setup DMA controller for playback */
++	if((err = configure_write_channel(&mxc_mc13783->s[SNDRV_PCM_STREAM_PLAYBACK],
++					audio_dma_irq)) < 0 )
++		return err;
++
++	if((prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL)) == NULL) {
++		ret = -ENOMEM;
++		goto out;
++	}
++
++	runtime->private_data = prtd;
++	return 0;
++
++ err1:
++	kfree(prtd);
++ out:
++	return ret;
++}
++
++static int mxc_pcm_close(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++
++//	mxc_mc13783_t *chip;
++	audio_stream_t *s;
++	device_data_t* device;
++	int ssi;
++
++	//chip = snd_pcm_substream_chip(substream);
++	s = &chip->s[substream->pstr->stream];
++	device = &s->stream_device;
++	ssi = device->ssi;
++
++	//disable_stereodac();
++
++	ssi_transmit_enable(ssi, false);
++	ssi_interrupt_disable(ssi, ssi_tx_dma_interrupt_enable);
++	ssi_tx_fifo_enable(ssi, ssi_fifo_0, false);
++	ssi_enable(ssi, false);
++
++	chip->s[substream->pstr->stream].stream = NULL;
++
++	return 0;
++}
++
++static int
++mxc_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 mxc_pcm_ops = {
++	.open		= mxc_pcm_open,
++	.close		= mxc_pcm_close,
++	.ioctl		= snd_pcm_lib_ioctl,
++	.hw_params	= mxc_pcm_hw_params,
++	.hw_free	= mxc_pcm_hw_free,
++	.prepare	= mxc_pcm_prepare,
++	.trigger	= mxc_pcm_trigger,
++	.pointer	= mxc_pcm_pointer,
++	.mmap		= mxc_pcm_mmap,
++};
++
++static u64 mxc_pcm_dmamask = 0xffffffff;
++
++int mxc_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 = &mxc_pcm_dmamask;
++	if (!card->dev->coherent_dma_mask)
++		card->dev->coherent_dma_mask = 0xffffffff;
++
++	if (dai->playback.channels_min) {
++		ret = mxc_pcm_preallocate_dma_buffer(pcm,
++			SNDRV_PCM_STREAM_PLAYBACK);
++		if (ret)
++			goto out;
++	}
++
++	if (dai->capture.channels_min) {
++		ret = mxc_pcm_preallocate_dma_buffer(pcm,
++			SNDRV_PCM_STREAM_CAPTURE);
++		if (ret)
++			goto out;
++	}
++ out:
++	return ret;
++}
++
++struct snd_soc_platform mxc_soc_platform = {
++	.name		= "mxc-audio",
++	.pcm_ops 	= &mxc_pcm_ops,
++	.pcm_new	= mxc_pcm_new,
++	.pcm_free	= mxc_pcm_free_dma_buffers,
++};
++
++EXPORT_SYMBOL_GPL(mxc_soc_platform);
++
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_DESCRIPTION("Freescale i.MX PCM DMA module");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,237 @@
++/*
++ * mxc-pcm.h :- ASoC platform header for Freescale i.MX
++ *
++ * 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 _MXC_PCM_H
++#define _MXC_PCM_H
++
++struct {
++	char *name;			/* stream identifier */
++	dma_channel_params dma_params;
++} mxc_pcm_dma_param;
++
++extern struct snd_soc_cpu_dai mxc_ssi_dai[3];
++
++/* platform data */
++extern struct snd_soc_platform mxc_soc_platform;
++extern struct snd_ac97_bus_ops mxc_ac97_ops;
++
++/* temp until imx-regs.h is up2date */
++#define SSI1_STX0   __REG(IMX_SSI1_BASE + 0x00)
++#define SSI1_STX0_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x00)
++#define SSI1_STX1   __REG(IMX_SSI1_BASE + 0x04)
++#define SSI1_STX1_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x04)
++#define SSI1_SRX0   __REG(IMX_SSI1_BASE + 0x08)
++#define SSI1_SRX0_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x08)
++#define SSI1_SRX1   __REG(IMX_SSI1_BASE + 0x0c)
++#define SSI1_SRX1_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x0c)
++#define SSI1_SCR    __REG(IMX_SSI1_BASE + 0x10)
++#define SSI1_SISR   __REG(IMX_SSI1_BASE + 0x14)
++#define SSI1_SIER   __REG(IMX_SSI1_BASE + 0x18)
++#define SSI1_STCR   __REG(IMX_SSI1_BASE + 0x1c)
++#define SSI1_SRCR   __REG(IMX_SSI1_BASE + 0x20)
++#define SSI1_STCCR  __REG(IMX_SSI1_BASE + 0x24)
++#define SSI1_SRCCR  __REG(IMX_SSI1_BASE + 0x28)
++#define SSI1_SFCSR  __REG(IMX_SSI1_BASE + 0x2c)
++#define SSI1_STR    __REG(IMX_SSI1_BASE + 0x30)
++#define SSI1_SOR    __REG(IMX_SSI1_BASE + 0x34)
++#define SSI1_SACNT  __REG(IMX_SSI1_BASE + 0x38)
++#define SSI1_SACADD __REG(IMX_SSI1_BASE + 0x3c)
++#define SSI1_SACDAT __REG(IMX_SSI1_BASE + 0x40)
++#define SSI1_SATAG  __REG(IMX_SSI1_BASE + 0x44)
++#define SSI1_STMSK  __REG(IMX_SSI1_BASE + 0x48)
++#define SSI1_SRMSK  __REG(IMX_SSI1_BASE + 0x4c)
++
++#define SSI2_STX0   __REG(IMX_SSI2_BASE + 0x00)
++#define SSI2_STX0_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x00)
++#define SSI2_STX1   __REG(IMX_SSI2_BASE + 0x04)
++#define SSI2_STX1_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x04)
++#define SSI2_SRX0   __REG(IMX_SSI2_BASE + 0x08)
++#define SSI2_SRX0_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x08)
++#define SSI2_SRX1   __REG(IMX_SSI2_BASE + 0x0c)
++#define SSI2_SRX1_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x0c)
++#define SSI2_SCR    __REG(IMX_SSI2_BASE + 0x10)
++#define SSI2_SISR   __REG(IMX_SSI2_BASE + 0x14)
++#define SSI2_SIER   __REG(IMX_SSI2_BASE + 0x18)
++#define SSI2_STCR   __REG(IMX_SSI2_BASE + 0x1c)
++#define SSI2_SRCR   __REG(IMX_SSI2_BASE + 0x20)
++#define SSI2_STCCR  __REG(IMX_SSI2_BASE + 0x24)
++#define SSI2_SRCCR  __REG(IMX_SSI2_BASE + 0x28)
++#define SSI2_SFCSR  __REG(IMX_SSI2_BASE + 0x2c)
++#define SSI2_STR    __REG(IMX_SSI2_BASE + 0x30)
++#define SSI2_SOR    __REG(IMX_SSI2_BASE + 0x34)
++#define SSI2_SACNT  __REG(IMX_SSI2_BASE + 0x38)
++#define SSI2_SACADD __REG(IMX_SSI2_BASE + 0x3c)
++#define SSI2_SACDAT __REG(IMX_SSI2_BASE + 0x40)
++#define SSI2_SATAG  __REG(IMX_SSI2_BASE + 0x44)
++#define SSI2_STMSK  __REG(IMX_SSI2_BASE + 0x48)
++#define SSI2_SRMSK  __REG(IMX_SSI2_BASE + 0x4c)
++
++#define SSI_SCR_CLK_IST        (1 << 9)
++#define SSI_SCR_TCH_EN         (1 << 8)
++#define SSI_SCR_SYS_CLK_EN     (1 << 7)
++#define SSI_SCR_I2S_MODE_NORM  (0 << 5)
++#define SSI_SCR_I2S_MODE_MSTR  (1 << 5)
++#define SSI_SCR_I2S_MODE_SLAVE (2 << 5)
++#define SSI_SCR_SYN            (1 << 4)
++#define SSI_SCR_NET            (1 << 3)
++#define SSI_SCR_RE             (1 << 2)
++#define SSI_SCR_TE             (1 << 1)
++#define SSI_SCR_SSIEN          (1 << 0)
++
++#define SSI_SISR_CMDAU         (1 << 18)
++#define SSI_SISR_CMDDU         (1 << 17)
++#define SSI_SISR_RXT           (1 << 16)
++#define SSI_SISR_RDR1          (1 << 15)
++#define SSI_SISR_RDR0          (1 << 14)
++#define SSI_SISR_TDE1          (1 << 13)
++#define SSI_SISR_TDE0          (1 << 12)
++#define SSI_SISR_ROE1          (1 << 11)
++#define SSI_SISR_ROE0          (1 << 10)
++#define SSI_SISR_TUE1          (1 << 9)
++#define SSI_SISR_TUE0          (1 << 8)
++#define SSI_SISR_TFS           (1 << 7)
++#define SSI_SISR_RFS           (1 << 6)
++#define SSI_SISR_TLS           (1 << 5)
++#define SSI_SISR_RLS           (1 << 4)
++#define SSI_SISR_RFF1          (1 << 3)
++#define SSI_SISR_RFF0          (1 << 2)
++#define SSI_SISR_TFE1          (1 << 1)
++#define SSI_SISR_TFE0          (1 << 0)
++
++#define SSI_SIER_RDMAE         (1 << 22)
++#define SSI_SIER_RIE           (1 << 21)
++#define SSI_SIER_TDMAE         (1 << 20)
++#define SSI_SIER_TIE           (1 << 19)
++#define SSI_SIER_CMDAU_EN      (1 << 18)
++#define SSI_SIER_CMDDU_EN      (1 << 17)
++#define SSI_SIER_RXT_EN        (1 << 16)
++#define SSI_SIER_RDR1_EN       (1 << 15)
++#define SSI_SIER_RDR0_EN       (1 << 14)
++#define SSI_SIER_TDE1_EN       (1 << 13)
++#define SSI_SIER_TDE0_EN       (1 << 12)
++#define SSI_SIER_ROE1_EN       (1 << 11)
++#define SSI_SIER_ROE0_EN       (1 << 10)
++#define SSI_SIER_TUE1_EN       (1 << 9)
++#define SSI_SIER_TUE0_EN       (1 << 8)
++#define SSI_SIER_TFS_EN        (1 << 7)
++#define SSI_SIER_RFS_EN        (1 << 6)
++#define SSI_SIER_TLS_EN        (1 << 5)
++#define SSI_SIER_RLS_EN        (1 << 4)
++#define SSI_SIER_RFF1_EN       (1 << 3)
++#define SSI_SIER_RFF0_EN       (1 << 2)
++#define SSI_SIER_TFE1_EN       (1 << 1)
++#define SSI_SIER_TFE0_EN       (1 << 0)
++
++#define SSI_STCR_TXBIT0        (1 << 9)
++#define SSI_STCR_TFEN1         (1 << 8)
++#define SSI_STCR_TFEN0         (1 << 7)
++#define SSI_STCR_TFDIR         (1 << 6)
++#define SSI_STCR_TXDIR         (1 << 5)
++#define SSI_STCR_TSHFD         (1 << 4)
++#define SSI_STCR_TSCKP         (1 << 3)
++#define SSI_STCR_TFSI          (1 << 2)
++#define SSI_STCR_TFSL          (1 << 1)
++#define SSI_STCR_TEFS          (1 << 0)
++
++#define SSI_SRCR_RXBIT0        (1 << 9)
++#define SSI_SRCR_RFEN1         (1 << 8)
++#define SSI_SRCR_RFEN0         (1 << 7)
++#define SSI_SRCR_RFDIR         (1 << 6)
++#define SSI_SRCR_RXDIR         (1 << 5)
++#define SSI_SRCR_RSHFD         (1 << 4)
++#define SSI_SRCR_RSCKP         (1 << 3)
++#define SSI_SRCR_RFSI          (1 << 2)
++#define SSI_SRCR_RFSL          (1 << 1)
++#define SSI_SRCR_REFS          (1 << 0)
++
++#define SSI_STCCR_DIV2         (1 << 18)
++#define SSI_STCCR_PSR          (1 << 15)
++#define SSI_STCCR_WL(x)        ((((x) - 2) >> 1) << 13)
++#define SSI_STCCR_DC(x)        (((x) & 0x1f) << 8)
++#define SSI_STCCR_PM(x)        (((x) & 0xff) << 0)
++
++#define SSI_SRCCR_DIV2         (1 << 18)
++#define SSI_SRCCR_PSR          (1 << 15)
++#define SSI_SRCCR_WL(x)        ((((x) - 2) >> 1) << 13)
++#define SSI_SRCCR_DC(x)        (((x) & 0x1f) << 8)
++#define SSI_SRCCR_PM(x)        (((x) & 0xff) << 0)
++
++
++#define SSI_SFCSR_RFCNT1(x)   (((x) & 0xf) << 28)
++#define SSI_SFCSR_TFCNT1(x)   (((x) & 0xf) << 24)
++#define SSI_SFCSR_RFWM1(x)    (((x) & 0xf) << 20)
++#define SSI_SFCSR_TFWM1(x)    (((x) & 0xf) << 16)
++#define SSI_SFCSR_RFCNT0(x)   (((x) & 0xf) << 12)
++#define SSI_SFCSR_TFCNT0(x)   (((x) & 0xf) <<  8)
++#define SSI_SFCSR_RFWM0(x)    (((x) & 0xf) <<  4)
++#define SSI_SFCSR_TFWM0(x)    (((x) & 0xf) <<  0)
++
++#define SSI_STR_TEST          (1 << 15)
++#define SSI_STR_RCK2TCK       (1 << 14)
++#define SSI_STR_RFS2TFS       (1 << 13)
++#define SSI_STR_RXSTATE(x)    (((x) & 0xf) << 8)
++#define SSI_STR_TXD2RXD       (1 <<  7)
++#define SSI_STR_TCK2RCK       (1 <<  6)
++#define SSI_STR_TFS2RFS       (1 <<  5)
++#define SSI_STR_TXSTATE(x)    (((x) & 0xf) << 0)
++
++#define SSI_SOR_CLKOFF        (1 << 6)
++#define SSI_SOR_RX_CLR        (1 << 5)
++#define SSI_SOR_TX_CLR        (1 << 4)
++#define SSI_SOR_INIT          (1 << 3)
++#define SSI_SOR_WAIT(x)       (((x) & 0x3) << 1)
++#define SSI_SOR_SYNRST        (1 << 0)
++
++#define SSI_SACNT_FRDIV(x)    (((x) & 0x3f) << 5)
++#define SSI_SACNT_WR          (x << 4)
++#define SSI_SACNT_RD          (x << 3)
++#define SSI_SACNT_TIF         (x << 2)
++#define SSI_SACNT_FV          (x << 1)
++#define SSI_SACNT_A97EN       (x << 0)
++
++
++/* AUDMUX registers */
++#define AUDMUX_HPCR1         __REG(IMX_AUDMUX_BASE + 0x00)
++#define AUDMUX_HPCR2         __REG(IMX_AUDMUX_BASE + 0x04)
++#define AUDMUX_HPCR3         __REG(IMX_AUDMUX_BASE + 0x08)
++#define AUDMUX_PPCR1         __REG(IMX_AUDMUX_BASE + 0x10)
++#define AUDMUX_PPCR2         __REG(IMX_AUDMUX_BASE + 0x14)
++#define AUDMUX_PPCR3         __REG(IMX_AUDMUX_BASE + 0x18)
++
++#define AUDMUX_HPCR_TFSDIR         (1 << 31)
++#define AUDMUX_HPCR_TCLKDIR        (1 << 30)
++#define AUDMUX_HPCR_TFCSEL_TX      (0 << 26)
++#define AUDMUX_HPCR_TFCSEL_RX      (8 << 26)
++#define AUDMUX_HPCR_TFCSEL(x)      (((x) & 0x7) << 26)
++#define AUDMUX_HPCR_RFSDIR         (1 << 25)
++#define AUDMUX_HPCR_RCLKDIR        (1 << 24)
++#define AUDMUX_HPCR_RFCSEL_TX      (0 << 20)
++#define AUDMUX_HPCR_RFCSEL_RX      (8 << 20)
++#define AUDMUX_HPCR_RFCSEL(x)      (((x) & 0x7) << 20)
++#define AUDMUX_HPCR_RXDSEL(x)      (((x) & 0x7) << 13)
++#define AUDMUX_HPCR_SYN            (1 << 12)
++#define AUDMUX_HPCR_TXRXEN         (1 << 10)
++#define AUDMUX_HPCR_INMEN          (1 <<  8)
++#define AUDMUX_HPCR_INMMASK(x)     (((x) & 0xff) << 0)
++
++#define AUDMUX_PPCR_TFSDIR         (1 << 31)
++#define AUDMUX_PPCR_TCLKDIR        (1 << 30)
++#define AUDMUX_PPCR_TFCSEL_TX      (0 << 26)
++#define AUDMUX_PPCR_TFCSEL_RX      (8 << 26)
++#define AUDMUX_PPCR_TFCSEL(x)      (((x) & 0x7) << 26)
++#define AUDMUX_PPCR_RFSDIR         (1 << 25)
++#define AUDMUX_PPCR_RCLKDIR        (1 << 24)
++#define AUDMUX_PPCR_RFCSEL_TX      (0 << 20)
++#define AUDMUX_PPCR_RFCSEL_RX      (8 << 20)
++#define AUDMUX_PPCR_RFCSEL(x)      (((x) & 0x7) << 20)
++#define AUDMUX_PPCR_RXDSEL(x)      (((x) & 0x7) << 13)
++#define AUDMUX_PPCR_SYN            (1 << 12)
++#define AUDMUX_PPCR_TXRXEN         (1 << 10)
++
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,417 @@
++/*
++ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's
++ *
++ * Copyright 2006 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ * Based on pxa2xx-pcm.c by	Nicolas Pitre, (C) 2004 MontaVista Software, Inc.
++ * and on mxc-alsa-mc13783 (C) 2006 Freescale.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License version 2 as
++ * published by the Free Software Foundation.
++ *
++ *  Revision history
++ *    29th Aug 2006   Initial version.
++ *
++ */
++
++#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/arch/dma.h>
++#include <asm/arch/spba.h>
++#include <asm/arch/clock.h>
++#include <asm/mach-types.h>
++#include <asm/hardware.h>
++
++#include "imx31-pcm.h"
++
++/* debug */
++#define IMX_DEBUG 0
++#if IMX_DEBUG
++#define dbg(format, arg...) printk(format, ## arg)
++#else
++#define dbg(format, arg...)
++#endif
++
++static const struct snd_pcm_hardware mxc_pcm_hardware = {
++	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
++				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
++				   SNDRV_PCM_INFO_MMAP |
++				   SNDRV_PCM_INFO_MMAP_VALID |
++				   SNDRV_PCM_INFO_PAUSE |
++				   SNDRV_PCM_INFO_RESUME),
++	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
++					SNDRV_PCM_FMTBIT_S24_LE,
++	.buffer_bytes_max	= 32 * 1024,
++	.period_bytes_min	= 64,
++	.period_bytes_max	= 8 * 1024,
++	.periods_min		= 2,
++	.periods_max		= 255,
++	.fifo_size		= 0,
++};
++
++struct mxc_runtime_data {
++	int dma_ch;
++	struct mxc_pcm_dma_param *dma_params;
++	spinlock_t dma_lock;
++	int active, period, periods;
++	int dma_wchannel;
++	int tx_spin, rx_spin;
++	int old_offset;
++};
++
++/*!
++  * This function stops the current dma transfer for playback
++  * and clears the dma pointers.
++  *
++  * @param	substream	pointer to the structure of the current stream.
++  *
++  */
++static void audio_stop_dma(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size);
++	unsigned int offset = dma_size * runtime->periods;
++	unsigned long flags;
++
++	spin_lock_irqsave(&prtd->dma_lock, flags);
++
++	dbg("MXC : audio_stop_dma active = 0\n");
++	prtd->active = 0;
++	prtd->period = 0;
++	prtd->periods = 0;
++
++	/* this stops the dma channel and clears the buffer ptrs */
++	mxc_dma_stop(prtd->dma_wchannel);
++	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_TO_DEVICE);
++	else
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_FROM_DEVICE);
++
++	spin_unlock_irqrestore(&prtd->dma_lock, flags);
++}
++
++/*!
++  * This function is called whenever a new audio block needs to be
++  * transferred to mc13783. The function receives the address and the size
++  * of the new block and start a new DMA transfer.
++  *
++  * @param	substream	pointer to the structure of the current stream.
++  *
++  */
++static int dma_new_period(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime =  substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	unsigned int dma_size;
++	unsigned int offset;
++	int ret = 0;
++	dma_request_t sdma_request;
++
++	if (prtd->active){
++		memset(&sdma_request, 0, sizeof(dma_request_t));
++		dma_size = frames_to_bytes(runtime, runtime->period_size);
++	    dbg("s->period (%x) runtime->periods (%d)\n",
++			s->period,runtime->periods);
++		dbg("runtime->period_size (%d) dma_size (%d)\n",
++			(unsigned int)runtime->period_size,
++			runtime->dma_bytes);
++
++       	offset = dma_size * prtd->period;
++//		snd_assert(dma_size <= DMA_BUF_SIZE, );
++		if(substream == SNDRV_PCM_STREAM_PLAYBACK)
++			sdma_request.sourceAddr = (char*)(dma_map_single(NULL,
++				runtime->dma_area + offset, dma_size, DMA_TO_DEVICE));
++		else
++			sdma_request.destAddr = (char*)(dma_map_single(NULL,
++				runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE));
++		sdma_request.count = dma_size;
++
++		dbg("MXC: Start DMA offset (%d) size (%d)\n", offset,
++						 runtime->dma_bytes);
++
++       	mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0);
++		if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) {
++			dbg("audio_process_dma: cannot queue DMA buffer\
++							(%i)\n", ret);
++			return ret;
++		}
++		prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */
++		prtd->period++;
++		prtd->period %= runtime->periods;
++    }
++	return ret;
++}
++
++
++/*!
++  * This is a callback which will be called
++  * when a TX transfer finishes. The call occurs
++  * in interrupt context.
++  *
++  * @param	dat	pointer to the structure of the current stream.
++  *
++  */
++static void audio_dma_irq(void *data)
++{
++	struct snd_pcm_substream *substream;
++	struct snd_pcm_runtime *runtime;
++	struct mxc_runtime_data *prtd;
++	unsigned int dma_size;
++	unsigned int previous_period;
++	unsigned int offset;
++
++	substream = data;
++	runtime = substream->runtime;
++	prtd = runtime->private_data;
++	previous_period  = prtd->periods;
++	dma_size = frames_to_bytes(runtime, runtime->period_size);
++	offset = dma_size * previous_period;
++
++	prtd->tx_spin = 0;
++	prtd->periods++;
++	prtd->periods %= runtime->periods;
++
++	/*
++	  * Give back to the CPU the access to the non cached memory
++	  */
++	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_TO_DEVICE);
++	else
++		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
++							DMA_FROM_DEVICE);
++	/*
++	  * If we are getting a callback for an active stream then we inform
++	  * the PCM middle layer we've finished a period
++	  */
++ 	if (prtd->active)
++		snd_pcm_period_elapsed(substream);
++
++	/*
++	  * Trig next DMA transfer
++	  */
++	dma_new_period(substream);
++}
++
++/*!
++  * This function configures the hardware to allow audio
++  * playback operations. It is called by ALSA framework.
++  *
++  * @param	substream	pointer to the structure of the current stream.
++  *
++  * @return              0 on success, -1 otherwise.
++  */
++static int
++mxc_pcm_prepare(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime =  substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	dma_channel_params *params = rtd->dai->cpu_dai->dma_data;
++	int ret = 0, channel = 0; // passed in ?;
++
++	prtd->period = 0;
++	prtd->periods = 0;
++
++	if(substream == SNDRV_PCM_STREAM_PLAYBACK) {
++		ret  = mxc_request_dma(&channel, "ALSA TX SDMA");
++		if (ret < 0) {
++			dbg("error requesting a write dma channel\n");
++			return ret;
++		}
++
++	} else {
++		ret = mxc_request_dma(&channel, "ALSA RX SDMA");
++		if (ret < 0) {
++			dbg("error requesting a read dma channel\n");
++			return ret;
++		}
++	}
++
++	/* set up chn with params */
++	params->callback = audio_dma_irq;
++	mxc_dma_setup_channel(channel, params);
++	prtd->dma_wchannel = channel;
++
++	return ret;
++}
++
++static int mxc_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	int ret;
++
++	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
++	if(ret < 0)
++		return ret;
++	runtime->dma_addr = virt_to_phys(runtime->dma_area);
++
++	return ret;
++}
++
++static int mxc_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++	return snd_pcm_lib_free_pages(substream);
++}
++
++static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct mxc_runtime_data *prtd = substream->runtime->private_data;
++	int ret = 0;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		prtd->tx_spin = 0;
++		/* requested stream startup */
++		prtd->active = 1;
++        ret = dma_new_period(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		/* requested stream shutdown */
++		audio_stop_dma(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++		prtd->active = 0;
++		prtd->periods = 0;
++		break;
++	case SNDRV_PCM_TRIGGER_RESUME:
++		prtd->active = 1;
++		prtd->tx_spin = 0;
++		ret = dma_new_period(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		prtd->active = 0;
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		prtd->active = 1;
++		if (prtd->old_offset) {
++			prtd->tx_spin = 0;
++            ret = dma_new_period(substream);
++		}
++		break;
++	default:
++		ret = -EINVAL;
++		break;
++	}
++
++	return ret;
++}
++
++static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++	unsigned int offset = 0;
++
++	/* tx_spin value is used here to check if a transfert is active */
++	if (prtd->tx_spin){
++		offset = (runtime->period_size * (prtd->periods)) +
++						(runtime->period_size >> 1);
++		if (offset >= runtime->buffer_size)
++			offset = runtime->period_size >> 1;
++	} else {
++		offset = (runtime->period_size * (prtd->periods));
++		if (offset >= runtime->buffer_size)
++			offset = 0;
++	}
++
++	return offset;
++}
++
++
++static int mxc_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd;
++	int ret;
++
++	snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware);
++
++	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
++	if (ret < 0)
++		return ret;
++	//ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
++	//	&hw_playback_rates);
++	//if (ret < 0)
++	//	return ret;
++
++	prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL);
++	if(prtd == NULL)
++		return -ENOMEM;
++
++	runtime->private_data = prtd;
++	return 0;
++}
++
++static int mxc_pcm_close(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct mxc_runtime_data *prtd = runtime->private_data;
++
++	kfree(prtd);
++	return 0;
++}
++
++static int
++mxc_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 mxc_pcm_ops = {
++	.open		= mxc_pcm_open,
++	.close		= mxc_pcm_close,
++	.ioctl		= snd_pcm_lib_ioctl,
++	.hw_params	= mxc_pcm_hw_params,
++	.hw_free	= mxc_pcm_hw_free,
++	.prepare	= mxc_pcm_prepare,
++	.trigger	= mxc_pcm_trigger,
++	.pointer	= mxc_pcm_pointer,
++	.mmap		= mxc_pcm_mmap,
++};
++
++static u64 mxc_pcm_dmamask = 0xffffffff;
++
++int mxc_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 = &mxc_pcm_dmamask;
++	if (!card->dev->coherent_dma_mask)
++		card->dev->coherent_dma_mask = 0xffffffff;
++
++	return ret;
++}
++
++struct snd_soc_platform mxc_soc_platform = {
++	.name		= "mxc-audio",
++	.pcm_ops 	= &mxc_pcm_ops,
++	.pcm_new	= mxc_pcm_new,
++};
++
++EXPORT_SYMBOL_GPL(mxc_soc_platform);
++
++MODULE_AUTHOR("Liam Girdwood");
++MODULE_DESCRIPTION("Freescale i.MX31 PCM DMA module");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,241 @@
++/*
++ * mxc-pcm.h :- ASoC platform header for Freescale i.MX
++ *
++ * 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 _MXC_PCM_H
++#define _MXC_PCM_H
++
++#include <asm/arch/dma.h>
++
++/* temp until imx-regs.h is up2date */
++#define SSI1_STX0   (SSI1_BASE_ADDR + 0x00)
++#define SSI1_STX0_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x00)
++#define SSI1_STX1   (SSI1_BASE_ADDR + 0x04)
++#define SSI1_STX1_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x04)
++#define SSI1_SRX0   (SSI1_BASE_ADDR + 0x08)
++#define SSI1_SRX0_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x08)
++#define SSI1_SRX1   (SSI1_BASE_ADDR + 0x0c)
++#define SSI1_SRX1_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x0c)
++#define SSI1_SCR    (SSI1_BASE_ADDR + 0x10)
++#define SSI1_SISR   (SSI1_BASE_ADDR + 0x14)
++#define SSI1_SIER   (SSI1_BASE_ADDR + 0x18)
++#define SSI1_STCR   (SSI1_BASE_ADDR + 0x1c)
++#define SSI1_SRCR   (SSI1_BASE_ADDR + 0x20)
++#define SSI1_STCCR  (SSI1_BASE_ADDR + 0x24)
++#define SSI1_SRCCR  (SSI1_BASE_ADDR + 0x28)
++#define SSI1_SFCSR  (SSI1_BASE_ADDR + 0x2c)
++#define SSI1_STR    (SSI1_BASE_ADDR + 0x30)
++#define SSI1_SOR    (SSI1_BASE_ADDR + 0x34)
++#define SSI1_SACNT  (SSI1_BASE_ADDR + 0x38)
++#define SSI1_SACADD (SSI1_BASE_ADDR + 0x3c)
++#define SSI1_SACDAT (SSI1_BASE_ADDR + 0x40)
++#define SSI1_SATAG  (SSI1_BASE_ADDR + 0x44)
++#define SSI1_STMSK  (SSI1_BASE_ADDR + 0x48)
++#define SSI1_SRMSK  (SSI1_BASE_ADDR + 0x4c)
++
++#define SSI2_STX0   (SSI2_BASE_ADDR + 0x00)
++#define SSI2_STX0_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x00)
++#define SSI2_STX1   (SSI2_BASE_ADDR + 0x04)
++#define SSI2_STX1_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x04)
++#define SSI2_SRX0   (SSI2_BASE_ADDR + 0x08)
++#define SSI2_SRX0_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x08)
++#define SSI2_SRX1   (SSI2_BASE_ADDR + 0x0c)
++#define SSI2_SRX1_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x0c)
++#define SSI2_SCR    (SSI2_BASE_ADDR + 0x10)
++#define SSI2_SISR   (SSI2_BASE_ADDR + 0x14)
++#define SSI2_SIER   (SSI2_BASE_ADDR + 0x18)
++#define SSI2_STCR   (SSI2_BASE_ADDR + 0x1c)
++#define SSI2_SRCR   (SSI2_BASE_ADDR + 0x20)
++#define SSI2_STCCR  (SSI2_BASE_ADDR + 0x24)
++#define SSI2_SRCCR  (SSI2_BASE_ADDR + 0x28)
++#define SSI2_SFCSR  (SSI2_BASE_ADDR + 0x2c)
++#define SSI2_STR    (SSI2_BASE_ADDR + 0x30)
++#define SSI2_SOR    (SSI2_BASE_ADDR + 0x34)
++#define SSI2_SACNT  (SSI2_BASE_ADDR + 0x38)
++#define SSI2_SACADD (SSI2_BASE_ADDR + 0x3c)
++#define SSI2_SACDAT (SSI2_BASE_ADDR + 0x40)
++#define SSI2_SATAG  (SSI2_BASE_ADDR + 0x44)
++#define SSI2_STMSK  (SSI2_BASE_ADDR + 0x48)
++#define SSI2_SRMSK  (SSI2_BASE_ADDR + 0x4c)
++
++#define SSI_SCR_CLK_IST        (1 << 9)
++#define SSI_SCR_TCH_EN         (1 << 8)
++#define SSI_SCR_SYS_CLK_EN     (1 << 7)
++#define SSI_SCR_I2S_MODE_NORM  (0 << 5)
++#define SSI_SCR_I2S_MODE_MSTR  (1 << 5)
++#define SSI_SCR_I2S_MODE_SLAVE (2 << 5)
++#define SSI_SCR_SYN            (1 << 4)
++#define SSI_SCR_NET            (1 << 3)
++#define SSI_SCR_RE             (1 << 2)
++#define SSI_SCR_TE             (1 << 1)
++#define SSI_SCR_SSIEN          (1 << 0)
++
++#define SSI_SISR_CMDAU         (1 << 18)
++#define SSI_SISR_CMDDU         (1 << 17)
++#define SSI_SISR_RXT           (1 << 16)
++#define SSI_SISR_RDR1          (1 << 15)
++#define SSI_SISR_RDR0          (1 << 14)
++#define SSI_SISR_TDE1          (1 << 13)
++#define SSI_SISR_TDE0          (1 << 12)
++#define SSI_SISR_ROE1          (1 << 11)
++#define SSI_SISR_ROE0          (1 << 10)
++#define SSI_SISR_TUE1          (1 << 9)
++#define SSI_SISR_TUE0          (1 << 8)
++#define SSI_SISR_TFS           (1 << 7)
++#define SSI_SISR_RFS           (1 << 6)
++#define SSI_SISR_TLS           (1 << 5)
++#define SSI_SISR_RLS           (1 << 4)
++#define SSI_SISR_RFF1          (1 << 3)
++#define SSI_SISR_RFF0          (1 << 2)
++#define SSI_SISR_TFE1          (1 << 1)
++#define SSI_SISR_TFE0          (1 << 0)
++
++#define SSI_SIER_RDMAE         (1 << 22)
++#define SSI_SIER_RIE           (1 << 21)
++#define SSI_SIER_TDMAE         (1 << 20)
++#define SSI_SIER_TIE           (1 << 19)
++#define SSI_SIER_CMDAU_EN      (1 << 18)
++#define SSI_SIER_CMDDU_EN      (1 << 17)
++#define SSI_SIER_RXT_EN        (1 << 16)
++#define SSI_SIER_RDR1_EN       (1 << 15)
++#define SSI_SIER_RDR0_EN       (1 << 14)
++#define SSI_SIER_TDE1_EN       (1 << 13)
++#define SSI_SIER_TDE0_EN       (1 << 12)
++#define SSI_SIER_ROE1_EN       (1 << 11)
++#define SSI_SIER_ROE0_EN       (1 << 10)
++#define SSI_SIER_TUE1_EN       (1 << 9)
++#define SSI_SIER_TUE0_EN       (1 << 8)
++#define SSI_SIER_TFS_EN        (1 << 7)
++#define SSI_SIER_RFS_EN        (1 << 6)
++#define SSI_SIER_TLS_EN        (1 << 5)
++#define SSI_SIER_RLS_EN        (1 << 4)
++#define SSI_SIER_RFF1_EN       (1 << 3)
++#define SSI_SIER_RFF0_EN       (1 << 2)
++#define SSI_SIER_TFE1_EN       (1 << 1)
++#define SSI_SIER_TFE0_EN       (1 << 0)
++
++#define SSI_STCR_TXBIT0        (1 << 9)
++#define SSI_STCR_TFEN1         (1 << 8)
++#define SSI_STCR_TFEN0         (1 << 7)
++#define SSI_STCR_TFDIR         (1 << 6)
++#define SSI_STCR_TXDIR         (1 << 5)
++#define SSI_STCR_TSHFD         (1 << 4)
++#define SSI_STCR_TSCKP         (1 << 3)
++#define SSI_STCR_TFSI          (1 << 2)
++#define SSI_STCR_TFSL          (1 << 1)
++#define SSI_STCR_TEFS          (1 << 0)
++
++#define SSI_SRCR_RXBIT0        (1 << 9)
++#define SSI_SRCR_RFEN1         (1 << 8)
++#define SSI_SRCR_RFEN0         (1 << 7)
++#define SSI_SRCR_RFDIR         (1 << 6)
++#define SSI_SRCR_RXDIR         (1 << 5)
++#define SSI_SRCR_RSHFD         (1 << 4)
++#define SSI_SRCR_RSCKP         (1 << 3)
++#define SSI_SRCR_RFSI          (1 << 2)
++#define SSI_SRCR_RFSL          (1 << 1)
++#define SSI_SRCR_REFS          (1 << 0)
++
++#define SSI_STCCR_DIV2         (1 << 18)
++#define SSI_STCCR_PSR          (1 << 15)
++#define SSI_STCCR_WL(x)        ((((x) - 2) >> 1) << 13)
++#define SSI_STCCR_DC(x)        (((x) & 0x1f) << 8)
++#define SSI_STCCR_PM(x)        (((x) & 0xff) << 0)
++
++#define SSI_SRCCR_DIV2         (1 << 18)
++#define SSI_SRCCR_PSR          (1 << 15)
++#define SSI_SRCCR_WL(x)        ((((x) - 2) >> 1) << 13)
++#define SSI_SRCCR_DC(x)        (((x) & 0x1f) << 8)
++#define SSI_SRCCR_PM(x)        (((x) & 0xff) << 0)
++
++
++#define SSI_SFCSR_RFCNT1(x)   (((x) & 0xf) << 28)
++#define SSI_SFCSR_TFCNT1(x)   (((x) & 0xf) << 24)
++#define SSI_SFCSR_RFWM1(x)    (((x) & 0xf) << 20)
++#define SSI_SFCSR_TFWM1(x)    (((x) & 0xf) << 16)
++#define SSI_SFCSR_RFCNT0(x)   (((x) & 0xf) << 12)
++#define SSI_SFCSR_TFCNT0(x)   (((x) & 0xf) <<  8)
++#define SSI_SFCSR_RFWM0(x)    (((x) & 0xf) <<  4)
++#define SSI_SFCSR_TFWM0(x)    (((x) & 0xf) <<  0)
++
++#define SSI_STR_TEST          (1 << 15)
++#define SSI_STR_RCK2TCK       (1 << 14)
++#define SSI_STR_RFS2TFS       (1 << 13)
++#define SSI_STR_RXSTATE(x)    (((x) & 0xf) << 8)
++#define SSI_STR_TXD2RXD       (1 <<  7)
++#define SSI_STR_TCK2RCK       (1 <<  6)
++#define SSI_STR_TFS2RFS       (1 <<  5)
++#define SSI_STR_TXSTATE(x)    (((x) & 0xf) << 0)
++
++#define SSI_SOR_CLKOFF        (1 << 6)
++#define SSI_SOR_RX_CLR        (1 << 5)
++#define SSI_SOR_TX_CLR        (1 << 4)
++#define SSI_SOR_INIT          (1 << 3)
++#define SSI_SOR_WAIT(x)       (((x) & 0x3) << 1)
++#define SSI_SOR_SYNRST        (1 << 0)
++
++#define SSI_SACNT_FRDIV(x)    (((x) & 0x3f) << 5)
++#define SSI_SACNT_WR          (x << 4)
++#define SSI_SACNT_RD          (x << 3)
++#define SSI_SACNT_TIF         (x << 2)
++#define SSI_SACNT_FV          (x << 1)
++#define SSI_SACNT_AC97EN      (x << 0)
++
++
++/* AUDMUX registers */
++#define AUDMUX_HPCR1         (IMX_AUDMUX_BASE + 0x00)
++#define AUDMUX_HPCR2         (IMX_AUDMUX_BASE + 0x04)
++#define AUDMUX_HPCR3         (IMX_AUDMUX_BASE + 0x08)
++#define AUDMUX_PPCR1         (IMX_AUDMUX_BASE + 0x10)
++#define AUDMUX_PPCR2         (IMX_AUDMUX_BASE + 0x14)
++#define AUDMUX_PPCR3         (IMX_AUDMUX_BASE + 0x18)
++
++#define AUDMUX_HPCR_TFSDIR         (1 << 31)
++#define AUDMUX_HPCR_TCLKDIR        (1 << 30)
++#define AUDMUX_HPCR_TFCSEL_TX      (0 << 26)
++#define AUDMUX_HPCR_TFCSEL_RX      (8 << 26)
++#define AUDMUX_HPCR_TFCSEL(x)      (((x) & 0x7) << 26)
++#define AUDMUX_HPCR_RFSDIR         (1 << 25)
++#define AUDMUX_HPCR_RCLKDIR        (1 << 24)
++#define AUDMUX_HPCR_RFCSEL_TX      (0 << 20)
++#define AUDMUX_HPCR_RFCSEL_RX      (8 << 20)
++#define AUDMUX_HPCR_RFCSEL(x)      (((x) & 0x7) << 20)
++#define AUDMUX_HPCR_RXDSEL(x)      (((x) & 0x7) << 13)
++#define AUDMUX_HPCR_SYN            (1 << 12)
++#define AUDMUX_HPCR_TXRXEN         (1 << 10)
++#define AUDMUX_HPCR_INMEN          (1 <<  8)
++#define AUDMUX_HPCR_INMMASK(x)     (((x) & 0xff) << 0)
++
++#define AUDMUX_PPCR_TFSDIR         (1 << 31)
++#define AUDMUX_PPCR_TCLKDIR        (1 << 30)
++#define AUDMUX_PPCR_TFCSEL_TX      (0 << 26)
++#define AUDMUX_PPCR_TFCSEL_RX      (8 << 26)
++#define AUDMUX_PPCR_TFCSEL(x)      (((x) & 0x7) << 26)
++#define AUDMUX_PPCR_RFSDIR         (1 << 25)
++#define AUDMUX_PPCR_RCLKDIR        (1 << 24)
++#define AUDMUX_PPCR_RFCSEL_TX      (0 << 20)
++#define AUDMUX_PPCR_RFCSEL_RX      (8 << 20)
++#define AUDMUX_PPCR_RFCSEL(x)      (((x) & 0x7) << 20)
++#define AUDMUX_PPCR_RXDSEL(x)      (((x) & 0x7) << 13)
++#define AUDMUX_PPCR_SYN            (1 << 12)
++#define AUDMUX_PPCR_TXRXEN         (1 << 10)
++
++#define SDMA_TXFIFO_WATERMARK				0x4
++#define SDMA_RXFIFO_WATERMARK				0x6
++
++struct mxc_pcm_dma_params {
++	char *name;			/* stream identifier */
++	dma_channel_params params;
++};
++
++extern struct snd_soc_cpu_dai mxc_ssi_dai[3];
++
++/* platform data */
++extern struct snd_soc_platform mxc_soc_platform;
++extern struct snd_ac97_bus_ops mxc_ac97_ops;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/magician.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/magician.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,577 @@
++/*
++ * SoC audio for HTC Magician
++ *
++ * Copyright (c) 2006 Philipp Zabel <philipp.zabel at gmail.com>
++ *
++ * based on spitz.c,
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *          Richard Purdie <richard at 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/timer.h>
++#include <linux/interrupt.h>
++#include <linux/platform_device.h>
++#include <linux/delay.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/scoop.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/hardware.h>
++#include <asm/arch/magician.h>
++#include <asm/arch/magician_cpld.h>
++#include <asm/mach-types.h>
++#include "../codecs/uda1380.h"
++#include "pxa2xx-pcm.h"
++#include "pxa2xx-i2s.h"
++#include "pxa2xx-ssp.h"
++
++#define MAGICIAN_HP_OFF    0
++#define MAGICIAN_HEADSET   1
++#define MAGICIAN_HP        2
++
++#define MAGICIAN_SPK_ON    0
++#define MAGICIAN_SPK_OFF   1
++
++#define MAGICIAN_MIC       0
++#define MAGICIAN_MIC_EXT   1
++#define MAGICIAN_BT_HS     2
++
++/*
++ * 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 GPIO53_SSP1SYSCLK_MD	(53 | GPIO_ALT_FN_2_OUT)
++
++static int magician_jack_func = MAGICIAN_HP_OFF;
++static int magician_spk_func = MAGICIAN_SPK_ON;
++static int magician_in_sel = MAGICIAN_MIC;
++
++extern struct platform_device magician_cpld;
++
++static void magician_ext_control(struct snd_soc_codec *codec)
++{
++	if (magician_spk_func == MAGICIAN_SPK_ON)
++		snd_soc_dapm_set_endpoint(codec, "Speaker", 1);
++	else
++		snd_soc_dapm_set_endpoint(codec, "Speaker", 0);
++
++	/* set up jack connection */
++	switch(magician_jack_func) {
++	case MAGICIAN_HP:
++		/* enable and unmute hp jack, disable mic bias */
++		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
++		break;
++	case MAGICIAN_HEADSET:
++		/* enable mic jack and bias, mute hp */
++		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
++		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++		break;
++	case MAGICIAN_HP_OFF:
++		/* jack removed, everything off */
++		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
++		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++		break;
++	}
++/*
++	switch(magician_in_sel) {
++	case MAGICIAN_IN_MIC:
++		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++	case MAGICIAN_IN_MIC_EXT:
++		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++	case MAGICIAN_IN_BT_HS:
++		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++	}
++*/
++	snd_soc_dapm_sync_endpoints(codec);
++}
++
++static int magician_startup(snd_pcm_substream_t *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 */
++	magician_ext_control(codec);
++
++	return 0;
++}
++
++/*
++ * Magician uses SSP port for playback.
++ */
++static int magician_playback_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int acps, acds, div4;
++	int ret = 0;
++
++	/*
++	 * Rate = SSPSCLK / (word size(16))
++	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
++	 */
++	switch(params_rate(params)) {
++	case 8000:
++		acps = 32842000;
++		acds = PXA2XX_SSP_CLK_AUDIO_DIV_32; /* wrong - 32 bits/sample */
++		div4 = PXA2XX_SSP_CLK_SCDB_4;
++		break;
++	case 11025:
++		acps = 5622000;
++		acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; /* 16 bits/sample, 1 slot */
++		div4 = PXA2XX_SSP_CLK_SCDB_4;
++		break;
++	case 22050:
++		acps = 5622000;
++		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
++		div4 = PXA2XX_SSP_CLK_SCDB_4;
++		break;
++	case 44100:
++		acps = 11345000;
++		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
++		div4 = PXA2XX_SSP_CLK_SCDB_4;
++		break;
++	case 48000:
++		acps = 12235000;
++		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
++		div4 = PXA2XX_SSP_CLK_SCDB_4;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set audio clock as clock source */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,
++		SND_SOC_CLOCK_OUT);
++	if (ret < 0)
++		return ret;
++
++	/* set the SSP audio system clock ACDS divider */
++	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
++	if (ret < 0)
++		return ret;
++
++	/* set the SSP audio system clock SCDB divider4 */
++	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
++	if (ret < 0)
++		return ret;
++
++	/* set SSP audio pll clock */
++	ret = cpu_dai->dai_ops.set_pll(cpu_dai, 0, 0, acps);
++	if (ret < 0)
++		return ret;
++
++	/* pH5 -- may need to set other SSP port config */
++
++	return 0;
++}
++
++/*
++ * We have to enable the SSP port early so the UDA1380 can flush
++ * it's register cache. The UDA1380 can only write it's interpolator and
++ * decimator registers when the link is running.
++ */
++static int magician_playback_prepare(struct snd_pcm_substream *substream)
++{
++	/* enable SSP clock - is this needed ? */
++	SSCR0_P(1) |= SSCR0_SSE;
++
++	/* FIXME: ENABLE I2S */
++	SACR0 |= SACR0_BCKD;
++	SACR0 |= SACR0_ENB;
++	pxa_set_cken(CKEN8_I2S, 1);
++#if 0
++	printk ("enabled I2S clock\n");
++	printk("SSP port enabled now\nSSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x SSACD %x\n",
++	SSCR0_P(1), SSCR1_P(1),
++	SSTO_P(1), SSPSP_P(1),
++	SSSR_P(1), SSACD_P(1));
++#endif
++	return 0;
++}
++
++static int magician_playback_hw_free(struct snd_pcm_substream *substream)
++{
++	/* FIXME: DISABLE I2S */
++	SACR0 &= ~SACR0_ENB;
++	SACR0 &= ~SACR0_BCKD;
++	pxa_set_cken(CKEN8_I2S, 0);
++	printk ("disabled I2S clock\n");
++	return 0;
++}
++
++/*
++ * Magician uses I2S for capture.
++ */
++static int magician_capture_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	int ret = 0;
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai,
++		SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
++		SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set the I2S system clock as output */
++	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
++		SND_SOC_CLOCK_OUT);
++	if (ret < 0)
++		return ret;
++
++	/* pH5 may need to set other I2S config */
++
++	return 0;
++}
++
++/*
++ * We have to enable the I2S port early so the UDA1380 can flush
++ * it's register cache. The UDA1380 can only write it's interpolator and
++ * decimator registers when the link is running.
++ */
++static int magician_capture_prepare(struct snd_pcm_substream *substream)
++{
++	SACR0 |= SACR0_ENB;
++	return 0;
++}
++
++static struct snd_soc_ops magician_capture_ops = {
++	.startup = magician_startup,
++	.hw_params = magician_capture_hw_params,
++	.prepare = magician_capture_prepare,
++};
++
++static struct snd_soc_ops magician_playback_ops = {
++	.startup = magician_startup,
++	.hw_params = magician_playback_hw_params,
++	.prepare = magician_playback_prepare,
++	.hw_free = magician_playback_hw_free,
++};
++
++static int magician_get_jack(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++{
++	ucontrol->value.integer.value[0] = magician_jack_func;
++	return 0;
++}
++
++static int magician_set_jack(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++	if (magician_jack_func == ucontrol->value.integer.value[0])
++		return 0;
++
++	magician_jack_func = ucontrol->value.integer.value[0];
++	magician_ext_control(codec);
++	return 1;
++}
++
++static int magician_get_spk(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++{
++	ucontrol->value.integer.value[0] = magician_spk_func;
++	return 0;
++}
++
++static int magician_set_spk(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++{
++	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
++
++	if (magician_spk_func == ucontrol->value.integer.value[0])
++		return 0;
++
++	magician_spk_func = ucontrol->value.integer.value[0];
++	magician_ext_control(codec);
++	return 1;
++}
++
++static int magician_get_input(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++{
++	ucontrol->value.integer.value[0] = magician_in_sel;
++	return 0;
++}
++
++static int magician_set_input(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++{
++	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
++
++	if (magician_in_sel == ucontrol->value.integer.value[0])
++		return 0;
++
++	magician_in_sel = ucontrol->value.integer.value[0];
++//	magician_ext_control(codec);
++	switch(magician_in_sel) {
++	case MAGICIAN_MIC:
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL0);
++		magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL1);
++		break;
++	case MAGICIAN_MIC_EXT:
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL0);
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL1);
++	}
++
++	return 1;
++}
++
++static int magician_spk_power(struct snd_soc_dapm_widget *w, int event)
++{
++	if (SND_SOC_DAPM_EVENT_ON(event))
++		magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
++	else
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
++	return 0;
++}
++
++static int magician_hp_power(struct snd_soc_dapm_widget *w, int event)
++{
++	if (SND_SOC_DAPM_EVENT_ON(event))
++		magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++	else
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++	return 0;
++}
++
++static int magician_mic_bias(struct snd_soc_dapm_widget* w, int event)
++{
++	if (SND_SOC_DAPM_EVENT_ON(event))
++		magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
++	else
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
++	return 0;
++}
++
++/* magician machine dapm widgets */
++static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
++	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
++	SND_SOC_DAPM_MIC("Mic Jack", magician_mic_bias),
++	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
++};
++
++/* magician machine audio_map */
++static const char* audio_map[][3] = {
++
++	/* headphone connected to VOUTLHP, VOUTRHP */
++	{"Headphone Jack", NULL, "VOUTLHP"},
++	{"Headphone Jack", NULL, "VOUTRHP"},
++
++	/* ext speaker connected to VOUTL, VOUTR  */
++	{"Speaker", NULL , "VOUTL"},
++	{"Speaker", NULL , "VOUTR"},
++
++	/* mic is connected to VINM */
++	{"VINM", NULL, "Mic Jack"},
++
++	/* line is connected to VINL, VINR */
++	{"VINL", NULL, "Line Jack"},
++	{"VINR", NULL, "Line Jack"},
++
++	{NULL, NULL, NULL},
++};
++
++static const char *jack_function[] = {"Off", "Headset", "Headphone"};
++static const char *spk_function[] = {"On", "Off"};
++static const char *input_select[] = {"Internal Mic", "External Mic"};
++static const struct soc_enum magician_enum[] = {
++	SOC_ENUM_SINGLE_EXT(4, jack_function),
++	SOC_ENUM_SINGLE_EXT(2, spk_function),
++	SOC_ENUM_SINGLE_EXT(2, input_select),
++};
++
++static const struct snd_kcontrol_new uda1380_magician_controls[] = {
++	SOC_ENUM_EXT("Jack Function", magician_enum[0], magician_get_jack, magician_set_jack),
++	SOC_ENUM_EXT("Speaker Function", magician_enum[1], magician_get_spk, magician_set_spk),
++	SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input, magician_set_input),
++};
++
++/*
++ * Logic for a uda1380 as connected on a HTC Magician
++ */
++static int magician_uda1380_init(struct snd_soc_codec *codec)
++{
++	int i, err;
++
++printk("magician_uda1380_init\n");
++
++	/* NC codec pins */
++	snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
++	snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
++
++	/* Add magician specific controls */
++	for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {
++		if ((err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&uda1380_magician_controls[i], codec, NULL))) < 0)
++			return err;
++	}
++
++	/* Add magician specific widgets */
++	for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
++	}
++
++	/* Set up magician specific audio path interconnects */
++	for(i = 0; audio_map[i][0] != NULL; i++) {
++		snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++	}
++
++	snd_soc_dapm_sync_endpoints(codec);
++	return 0;
++}
++
++/* magician digital audio interface glue - connects codec <--> CPU */
++static struct snd_soc_dai_link magician_dai[] = {
++{
++	.name = "uda1380",
++	.stream_name = "UDA1380 Playback",
++	.cpu_dai = &pxa_ssp_dai[0],
++	.codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
++	.init = magician_uda1380_init,
++	.ops = &magician_playback_ops,
++},
++{
++	.name = "uda1380",
++	.stream_name = "UDA1380 Capture",
++	.cpu_dai = &pxa_i2s_dai,
++	.codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
++	.ops = &magician_capture_ops,
++}
++};
++
++/* magician audio machine driver */
++static struct snd_soc_machine snd_soc_machine_magician = {
++	.name = "Magician",
++	.dai_link = magician_dai,
++	.num_links = ARRAY_SIZE(magician_dai),
++};
++
++/* magician audio private data */
++static struct uda1380_setup_data magician_uda1380_setup = {
++	.i2c_address = 0x18,
++	.dac_clk = UDA1380_DAC_CLK_WSPLL,
++};
++
++/* magician audio subsystem */
++static struct snd_soc_device magician_snd_devdata = {
++	.machine = &snd_soc_machine_magician,
++	.platform = &pxa2xx_soc_platform,
++	.codec_dev = &soc_codec_dev_uda1380,
++	.codec_data = &magician_uda1380_setup,
++};
++
++static struct platform_device *magician_snd_device;
++
++static irqreturn_t magician_ep_plug_isr (int isr, void *data)
++{
++	//struct snd_soc_codec *codec = data;
++	int ep_state = magician_ep_state(&magician_cpld);
++
++	printk("earphone: %d\n", ep_state);
++
++	if (ep_state) {
++		//snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
++		magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++	} else {
++		magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++	}
++
++	return IRQ_HANDLED;
++}
++
++
++static int __init magician_init(void)
++{
++	int ret;
++
++	if (!machine_is_magician())
++		return -ENODEV;
++
++printk("snd-soc-magician init\n");
++
++	magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
++
++// we probably need to have the clock running here:
++
++	magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
++	udelay(5);
++	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
++
++	/* correct place? we'll need it to talk to the uda1380 */
++	request_module("i2c-pxa");
++
++	magician_snd_device = platform_device_alloc("soc-audio", -1);
++	if (!magician_snd_device)
++		return -ENOMEM;
++
++	platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
++	magician_snd_devdata.dev = &magician_snd_device->dev;
++	ret = platform_device_add(magician_snd_device);
++
++	if (ret)
++		platform_device_put(magician_snd_device);
++	request_irq(IRQ_MAGICIAN_EP, magician_ep_plug_isr, IRQF_SAMPLE_RANDOM,
++		     "EP plug", NULL);
++	/*&magician_snd_devdata->codec_dev->codec->parent(snd_soc_device)->codec*/
++
++	pxa_gpio_mode(GPIO53_SSP1SYSCLK_MD);
++	pxa_gpio_mode(GPIO26_SSP1RX_MD);
++	pxa_gpio_mode(GPIO25_SSP1TX_MD);
++	pxa_gpio_mode(GPIO23_SSP1CLKM_MD);
++	pxa_gpio_mode(GPIO24_SSP1FRMM_MD);
++
++	return ret;
++}
++
++static void __exit magician_exit(void)
++{
++	platform_device_unregister(magician_snd_device);
++	free_irq(IRQ_MAGICIAN_EP, NULL);
++
++	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
++	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
++	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
++}
++
++module_init(magician_init);
++module_exit(magician_exit);
++
++MODULE_AUTHOR("Philipp Zabel");
++MODULE_DESCRIPTION("ALSA SoC Magician");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-ac97.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-ac97.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,270 @@
++/*
++ * ltemplate-ac97.c -- AC97 support for the xxx chip.
++ *
++ * 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 <linux/mutex.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 <asm/hardware.h>
++
++#include "template-pcm.h"
++
++#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)
++
++/* DAI description of AC97 controllers capabilities */
++static struct snd_soc_dai_mode template_ac97_modes[] = {
++	{
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = AC97_RATES,
++		.pcmdir = AC97_DIR,
++	},
++};
++
++/* AC97 controlller reads codec register */
++static unsigned short template_ac97_read(struct snd_ac97 *ac97,
++	unsigned short reg)
++{
++}
++
++/* AC97 controller writes to codec register */
++static void template_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
++	unsigned short val)
++{
++}
++
++/* AC97 controller asserts a warm reset */
++static void template_ac97_warm_reset(struct snd_ac97 *ac97)
++{
++}
++
++/* AC97 controller asserts a cold reset */
++static void template_ac97_cold_reset(struct snd_ac97 *ac97)
++{
++}
++
++/* AC97 controller operations */
++struct snd_ac97_bus_ops soc_ac97_ops = {
++	.read	= template_ac97_read,
++	.write	= template_ac97_write,
++	.warm_reset	= template_ac97_warm_reset,
++	.reset	= template_ac97_cold_reset,
++};
++EXPORT_SYMBOL_GPL(soc_ac97_ops);
++
++/* DMA structure describing platform specific AC97 DMA for each logical DAI */
++static struct template_pcm_dma_params template_ac97_pcm_stereo_out = {
++	.name			= "AC97 PCM Stereo out",
++	.dev_addr		= __PREG(PCDR),
++};
++
++static struct template_pcm_dma_params template_ac97_pcm_stereo_in = {
++	.name			= "AC97 PCM Stereo in",
++	.dev_addr		= __PREG(PCDR),
++};
++
++static struct template_pcm_dma_params template_ac97_pcm_aux_mono_out = {
++	.name			= "AC97 Aux PCM (Slot 5) Mono out",
++	.dev_addr		= __PREG(MODR),
++};
++
++static struct template_pcm_dma_params template_ac97_pcm_aux_mono_in = {
++	.name			= "AC97 Aux PCM (Slot 5) Mono in",
++	.dev_addr		= __PREG(MODR),
++};
++
++static struct template_pcm_dma_params template_ac97_pcm_mic_mono_in = {
++	.name			= "AC97 Mic PCM (Slot 6) Mono in",
++	.dev_addr		= __PREG(MCDR),
++};
++
++#ifdef CONFIG_PM
++/* suspend the AC97 controller */
++static int template_ac97_suspend(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++}
++
++/* resume the AC97 controller */
++static int template_ac97_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++}
++
++#else
++#define template_ac97_suspend	NULL
++#define template_ac97_resume	NULL
++#endif
++
++/*
++ * Probe initialises the AC97 controller. e.g.
++ * request any IRQ's
++ * configure GPIO's
++ * enable any clocks
++ */
++static int template_ac97_probe(struct platform_device *pdev)
++{
++}
++
++/*
++ * Free's resources setup in probe()
++ */
++static void template_ac97_remove(struct platform_device *pdev)
++{
++}
++
++/*
++ * Alsa operations
++ * Only implement the required operations for your platform.
++ * These operations are specific to the AC97 controller and DAI only.
++ */
++
++ /*
++ * Called by ALSA when a PCM substream is opened, private data can be allocated.
++ */
++static int template_ac97_startup(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when a PCM substream is closed. Private data can be
++ * freed here.
++ */
++static int template_ac97_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the PCM substream is prepared, can set format, sample
++ * rate, etc.  This function is non atomic and can be called multiple times,
++ * it can refer to the runtime info.
++ */
++static int template_ac97_prepare(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the hardware params are set by application. This
++ * function can also be called multiple times and can allocate buffers
++ * (using snd_pcm_lib_* ). It's non-atomic.
++ */
++static int template_ac97_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++}
++
++/*
++ * Free's resources allocated by hw_params, can be called multiple times
++ */
++static int template_ac97_hw_free(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Starts (Triggers) audio playback or capture.
++ * Usually only needed for DMA
++ */
++static int template_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++}
++
++/*
++ * Define each AC97 slot grouping as a DAI.
++ *
++ * e.g. (below)
++ *  Slots 3 & 4 = HiFi DAI
++ *  Slot 5 = Aux playback
++ *  Slot 7 = Mic Capture
++ *
++ * This gives 3 logical DAI's on the 1 physical AC97 DAI.
++ *
++ */
++struct snd_soc_cpu_dai template_ac97_dai[] = {
++{
++	.name = "template-ac97-HiFi",
++	.id = 0,
++	.type = SND_SOC_DAI_AC97,
++	/* DAI driver operations - only needed on 1st logical DAI in AC97 */
++	.probe = template_ac97_probe,
++	.remove = template_ac97_remove,
++	.suspend = template_ac97_suspend,
++	.resume = template_ac97_resume,
++	/* playback and capture stream info */
++	.playback = {
++		.stream_name = "AC97 Playback",
++		.channels_min = 2,
++		.channels_max = 2,},
++	.capture = {
++		.stream_name = "AC97 Capture",
++		.channels_min = 2,
++		.channels_max = 2,},
++	/* alsa PCM operations */
++	.ops = {
++		.startup = template_ac97_startup,
++		.shutdown = template_ac97_shutdown,
++		.prepare = template_ac97_prepare,
++		.trigger = template_ac97_trigger,
++		.hw_params = template_ac97_hw_params,
++		.hw_free = template_ac97_hw_free,},
++	/* DAI capabilities */
++	.caps = {
++		.num_modes = ARRAY_SIZE(template_ac97_modes),
++		.mode = template_ac97_modes,},
++},
++/* AC97 AUX playback - not supported on all controllers */
++{
++	.name = "template-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 = template_ac97_hw_aux_params,},
++	.caps = {
++		.num_modes = ARRAY_SIZE(template_ac97_modes),
++		.mode = template_ac97_modes,},
++},
++/* AC97 Mic capture - not supported on all controllers */
++{
++	.name = "template-ac97-mic",
++	.id = 2,
++	.type = SND_SOC_DAI_AC97,
++	.capture = {
++		.stream_name = "AC97 Mic Capture",
++		.channels_min = 1,
++		.channels_max = 1,},
++	.ops = {
++		.hw_params = template_ac97_hw_mic_params,},
++	.caps = {
++		.num_modes = ARRAY_SIZE(template_ac97_modes),
++		.mode = template_ac97_modes,},},
++};
++EXPORT_SYMBOL_GPL(template_ac97_dai);
++
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,784 @@
++/*
++ * template-codec.c  --  Template Codec  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.
++ */
++
++#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 "template-codec.h"
++
++#define AUDIO_NAME "template-codec"
++#define TEMPLATE_VERSION "0.1"
++
++/*
++ * Debug
++ */
++
++#define TEMPLATE_DEBUG 0
++
++#ifdef TEMPLATE_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_template_codec;
++
++/*
++ * template_codec register cache
++ */
++static const u16 template_codec_reg[TEMPLATE_CACHEREGNUM] = {
++    0x0097, 0x0097, 0x0079, 0x0079,
++    0x000a, 0x0008, 0x009f, 0x000a,
++    0x0000, 0x0000
++};
++
++/* Codec DAI can support these hardware formats */
++#define TEMPLATE_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)
++
++/* Codec DAI supports direction */
++#define TEMPLATE_DIR \
++	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++/* Codec DAI supports rates */
++#define TEMPLATE_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)
++
++/* Codec DAI supports PCM word sizes */
++#define TEMPLATE_HIFI_BITS \
++	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++/*
++ * Description of supported codec DAI supported modes.
++ */
++static struct snd_soc_dai_mode template_codec_modes[] = {
++	/* codec frame and clock master modes */
++	/* 8k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_8000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 1536,
++		.bfs = 64,
++	},
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_8000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 2304,
++		.bfs = 64,
++	},
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_8000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 1408,
++		.bfs = 64,
++	},
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_8000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 2112,
++		.bfs = 64,
++	},
++
++	/* 32k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_32000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 384,
++		.bfs = 64,
++	},
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_32000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 576,
++		.bfs = 64,
++	},
++
++	/* 44.1k & 48k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 256,
++		.bfs = 64,
++	},
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 384,
++		.bfs = 64,
++	},
++
++	/* 88.2 & 96k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 128,
++		.bfs = 64,
++	},
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.fs = 192,
++		.bfs = 64,
++	},
++
++	/* USB codec frame and clock master modes */
++	/* 8k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_8000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 1500,
++		.bfs = SND_SOC_FSBD(1),
++	},
++
++	/* 44.1k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_44100,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 272,
++		.bfs = SND_SOC_FSBD(1),
++	},
++
++	/* 48k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_48000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 250,
++		.bfs = SND_SOC_FSBD(1),
++	},
++
++	/* 88.2k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_88200,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 136,
++		.bfs = SND_SOC_FSBD(1),
++	},
++
++	/* 96k */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = SNDRV_PCM_RATE_96000,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 125,
++		.bfs = SND_SOC_FSBD(1),
++	},
++
++	/* codec frame and clock slave modes */
++	{
++		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = TEMPLATE_HIFI_BITS,
++		.pcmrate = TEMPLATE_RATES,
++		.pcmdir = TEMPLATE_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = SND_SOC_FS_ALL,
++		.bfs = SND_SOC_FSB_ALL,
++	},
++};
++
++/*
++ * read template_codec register cache
++ */
++static inline unsigned int template_codec_read_reg_cache(struct snd_soc_codec *codec,
++	unsigned int reg)
++{
++	u16 *cache = codec->reg_cache;
++
++	if (reg >= TEMPLATE_CACHEREGNUM)
++		return -1;
++	return cache[reg];
++}
++
++/*
++ * write template_codec register cache
++ */
++static inline void template_codec_write_reg_cache(struct snd_soc_codec *codec,
++	u16 reg, unsigned int value)
++{
++	u16 *cache = codec->reg_cache;
++	if (reg >= TEMPLATE_CACHEREGNUM)
++		return;
++	cache[reg] = value;
++}
++
++/*
++ * write to the template codec register space
++ */
++static int template_codec_write(struct snd_soc_codec *codec, unsigned int reg,
++	unsigned int value)
++{
++	u8 data[2];
++
++	/* format the data - codec specific */
++	data[0] = reg;
++	data[1] = value;
++
++	template_codec_write_reg_cache (codec, reg, value);
++	if (codec->hw_write(codec->control_data, data, 2) == 2)
++		return 0;
++	else
++		return -EIO;
++}
++
++#define template_codec_reset(c)	template_codec_write(c, TEMPLATE_RESET, 0)
++
++
++/* template codec non DAPM controls */
++static const struct snd_kcontrol_new template_codec_snd_controls[] = {
++};
++
++/* add non dapm controls */
++static int template_codec_add_controls(struct snd_soc_codec *codec)
++{
++	int err, i;
++
++	for (i = 0; i < ARRAY_SIZE(template_codec_snd_controls); i++) {
++		if ((err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&template_codec_snd_controls[i],codec, NULL))) < 0)
++			return err;
++	}
++
++	return 0;
++}
++
++/* template codec DAPM controls */
++static const struct snd_soc_dapm_widget template_codec_dapm_widgets[] = {
++};
++
++/*
++ * template codec audio interconnectiosn between sink and source.
++ */
++static const char *audio_map[][3] = {
++
++
++	/* terminator */
++	{NULL, NULL, NULL},
++};
++
++static int template_codec_add_widgets(struct snd_soc_codec *codec)
++{
++	int i;
++
++	for(i = 0; i < ARRAY_SIZE(template_codec_dapm_widgets); i++) {
++		snd_soc_dapm_new_control(codec, &template_codec_dapm_widgets[i]);
++	}
++
++	/* set up audio path interconnects */
++	for(i = 0; intercon[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;
++}
++
++/*
++ * Configures the codec SYSCLK/MCLK (system or master clock)
++ */
++static unsigned int template_codec_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;
++}
++
++/*
++ * Alsa operations
++ * Only implement the required operations for your platform.
++ * These operations are specific to the codec only.
++ */
++
++ /*
++ * Called by ALSA when a PCM substream is opened, private data can be allocated.
++ */
++static int template_codec_startup(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when a PCM substream is closed. Private data can be
++ * freed here.
++ */
++static int template_codec_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the hardware params are set by application. This
++ * function can also be called multiple times and can allocate buffers
++ * (using snd_pcm_lib_* ). It's non-atomic.
++ */
++static int template_codec_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++}
++
++/*
++ * Free's resources allocated by hw_params, can be called multiple times
++ */
++static int template_codec_hw_free(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Starts (Triggers) audio playback or capture.
++ * Usually only needed for DMA
++ */
++static int template_codec_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++}
++
++/*
++ * Called by ALSA when the PCM substream is prepared, can set format, sample
++ * rate, etc.  This function is non atomic and can be called multiple times,
++ * it can refer to the runtime info.
++ */
++static int template_codec_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;
++
++	/* set master/slave audio interface */
++	switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		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:
++		break;
++	case SND_SOC_DAIFMT_RIGHT_J:
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		break;
++	case SND_SOC_DAIFMT_DSP_A:
++		break;
++	case SND_SOC_DAIFMT_DSP_B:
++		break;
++	}
++
++	/* bit size */
++	switch (rtd->codec_dai->dai_runtime.pcmfmt) {
++	case SNDRV_PCM_FMTBIT_S16_LE:
++		break;
++	case SNDRV_PCM_FMTBIT_S20_3LE:
++		break;
++	case SNDRV_PCM_FMTBIT_S24_LE:
++		break;
++	case SNDRV_PCM_FMTBIT_S32_LE:
++		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:
++		break;
++	case SND_SOC_DAIFMT_IB_NF:
++		break;
++	case SND_SOC_DAIFMT_NB_IF:
++		break;
++	}
++
++	return 0;
++}
++
++/*
++ * Enable / Disable codec digital soft mute
++ */
++static int template_codec_mute(struct snd_soc_codec *codec,
++	struct snd_soc_codec_dai *dai, int mute)
++{
++}
++
++/*
++ * Codec DAPM event handler
++ * This handles codec level DAPM events
++ */
++static int template_codec_dapm_event(struct snd_soc_codec *codec, int event)
++{
++	u16 reg = template_codec_read_reg_cache(codec, TEMPLATE_PWR) & 0xff7f;
++
++	switch (event) {
++	case SNDRV_CTL_POWER_D0: /* full On */
++		/* e.g. vref/mid, osc on, */
++		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, */
++		break;
++	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++		/* everything off, dac mute, inactive */
++		break;
++	}
++	codec->dapm_state = event;
++	return 0;
++}
++
++/*
++ * Define codec DAI.
++ */
++struct snd_soc_codec_dai template_codec_dai = {
++	.name = "codec xxx",
++	/* playback and capture stream info */
++	.playback = {
++		.stream_name = "Playback",
++		.channels_min = 1,
++		.channels_max = 2,
++	},
++	.capture = {
++		.stream_name = "Capture",
++		.channels_min = 1,
++		.channels_max = 2,
++	},
++	/* codec operations */
++	.config_sysclk = template_codec_config_sysclk,
++	.digital_mute = template_codec_mute,
++	/* alsa PCM operations */
++	.ops = {
++		.startup = template_codec_startup,
++		.shutdown = template_codec_shutdown,
++		.prepare = template_codec_prepare,
++		.trigger = template_codec_trigger,
++		.hw_params = template_codec_hw_params,
++		.hw_free = template_codec_hw_free,},
++	/* codec capabilities */
++	.caps = {
++		.num_modes = ARRAY_SIZE(template_codec_modes),
++		.mode = template_codec_modes,
++	},
++};
++EXPORT_SYMBOL_GPL(template_codec_dai);
++
++static int template_codec_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;
++
++	template_codec_write(codec, TEMPLATE_ACTIVE, 0x0);
++	template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
++	return 0;
++}
++
++static int template_codec_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(template_codec_reg); i++) {
++		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
++		data[1] = cache[i] & 0x00ff;
++		codec->hw_write(codec->control_data, data, 2);
++	}
++	template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++	template_codec_dapm_event(codec, codec->suspend_dapm_state);
++	return 0;
++}
++
++/*
++ * initialise the TEMPLATE driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int template_codec_init(struct snd_soc_device *socdev)
++{
++	struct snd_soc_codec *codec = socdev->codec;
++	int reg, ret = 0;
++
++	codec->name = "TEMPLATE";
++	codec->owner = THIS_MODULE;
++	codec->read = template_codec_read_reg_cache;
++	codec->write = template_codec_write;
++	codec->dapm_event = template_codec_dapm_event;
++	codec->dai = &template_codec_dai;
++	codec->num_dai = 1;
++	codec->reg_cache_size = ARRAY_SIZE(template_codec_reg);
++
++	codec->reg_cache =
++			kzalloc(sizeof(u16) * ARRAY_SIZE(template_codec_reg), GFP_KERNEL);
++	if (codec->reg_cache == NULL)
++		return -ENOMEM;
++	memcpy(codec->reg_cache,
++		template_codec_reg, sizeof(u16) * ARRAY_SIZE(template_codec_reg));
++	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(template_codec_reg);
++
++	template_codec_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 */
++	template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
++
++	/* set the update bits */
++	reg = template_codec_read_reg_cache(codec, TEMPLATE_LOUT1V);
++	template_codec_write(codec, TEMPLATE_LOUT1V, reg | 0x0100);
++	reg = template_codec_read_reg_cache(codec, TEMPLATE_ROUT1V);
++	template_codec_write(codec, TEMPLATE_ROUT1V, reg | 0x0100);
++	reg = template_codec_read_reg_cache(codec, TEMPLATE_LINVOL);
++	template_codec_write(codec, TEMPLATE_LINVOL, reg | 0x0100);
++	reg = template_codec_read_reg_cache(codec, TEMPLATE_RINVOL);
++	template_codec_write(codec, TEMPLATE_RINVOL, reg | 0x0100);
++
++	template_codec_add_controls(codec);
++	template_codec_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 *template_codec_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++/*
++ * TEMPLATE 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 template_codec_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 template_codec_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++	struct snd_soc_device *socdev = template_codec_socdev;
++	struct template_codec_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 = template_codec_init(socdev);
++	if (ret < 0) {
++		err("failed to initialise TEMPLATE\n");
++		goto err;
++	}
++	return ret;
++
++err:
++	kfree(codec);
++	kfree(i2c);
++	return ret;
++}
++
++static int template_codec_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 template_codec_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, template_codec_codec_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver template_codec_i2c_driver = {
++	.driver = {
++		.name = "TEMPLATE I2C Codec",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_TEMPLATE,
++	.attach_adapter = template_codec_i2c_attach,
++	.detach_client =  template_codec_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "TEMPLATE",
++	.driver = &template_codec_i2c_driver,
++};
++#endif
++
++static int template_codec_probe(struct platform_device *pdev)
++{
++	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++	struct template_codec_setup_data *setup;
++	struct snd_soc_codec *codec;
++	int ret = 0;
++
++	info("TEMPLATE Audio Codec %s", TEMPLATE_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);
++
++	template_codec_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(&template_codec_i2c_driver);
++		if (ret != 0)
++			printk(KERN_ERR "can't add i2c driver");
++	}
++#else
++	/* Add other interfaces here */
++#endif
++	return ret;
++}
++
++/* power down chip and remove */
++static int template_codec_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)
++		template_codec_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(&template_codec_i2c_driver);
++#endif
++	kfree(codec);
++
++	return 0;
++}
++
++/* codec device ops */
++struct snd_soc_codec_device soc_codec_dev_template_codec = {
++	.probe = 	template_codec_probe,
++	.remove = 	template_codec_remove,
++	.suspend = 	template_codec_suspend,
++	.resume =	template_codec_resume,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_template_codec);
++
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-i2s.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-i2s.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,223 @@
++/*
++ *  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/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 "template-pcm.h"
++
++/* supported I2S DAI hardware formats */
++#define TEMPLATE_I2S_DAIFMT \
++	(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF)
++
++/* supported I2S direction */
++#define TEMPLATE_I2S_DIR \
++	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
++
++/* supported I2S rates */
++#define TEMPLATE_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)
++
++/* I2S controller DAI capabilities */
++static struct snd_soc_dai_mode template_i2s_modes[] = {
++	/* template I2S frame and clock master modes */
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = SNDRV_PCM_RATE_8000,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 256,
++		.bfs = SND_SOC_FSBD(4),
++	},
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = SNDRV_PCM_RATE_11025,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 256,
++		.bfs = SND_SOC_FSBD(4),
++	},
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = SNDRV_PCM_RATE_16000,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 256,
++		.bfs = SND_SOC_FSBD(4),
++	},
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = SNDRV_PCM_RATE_22050,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 256,
++		.bfs = SND_SOC_FSBD(4),
++	},
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = SNDRV_PCM_RATE_44100,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 256,
++		.bfs = SND_SOC_FSBD(4),
++	},
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = SNDRV_PCM_RATE_48000,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.flags = SND_SOC_DAI_BFS_DIV,
++		.fs = 256,
++		.bfs = SND_SOC_FSBD(4),
++	},
++
++	/* template I2S frame master and clock slave mode */
++	{
++		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS,
++		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
++		.pcmrate = TEMPLATE_I2S_RATES,
++		.pcmdir = TEMPLATE_I2S_DIR,
++		.fs = SND_SOC_FS_ALL,
++		.flags = SND_SOC_DAI_BFS_RATE,
++		.bfs = 64,
++	},
++};
++
++/* I2S controller platform specific DMA parameters */
++static struct template_pcm_dma_params template_i2s_pcm_stereo_out = {
++	.name			= "I2S PCM Stereo out",
++	.dev_addr		= __PREG(SADR),
++};
++
++static struct template_pcm_dma_params template_i2s_pcm_stereo_in = {
++	.name			= "I2S PCM Stereo in",
++	.dev_addr		= __PREG(SADR),
++};
++
++#ifdef CONFIG_PM
++/* suspend I2S controller */
++static int template_i2s_suspend(struct platform_device *dev,
++	struct snd_soc_cpu_dai *dai)
++{
++}
++
++/* resume I2S controller */
++static int template_i2s_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++}
++
++#else
++#define template_i2s_suspend	NULL
++#define template_i2s_resume	NULL
++#endif
++
++/* configure the I2S controllers MCLK or SYSCLK */
++static unsigned int template_i2s_config_sysclk(struct snd_soc_cpu_dai *iface,
++	struct snd_soc_clock_info *info, unsigned int clk)
++{
++}
++
++
++/*
++ * Alsa operations
++ * Only implement the required operations for your platform.
++ * These operations are specific to the I2S controller and DAI only.
++ */
++
++ /*
++ * Called by ALSA when a PCM substream is opened, private data can be allocated.
++ */
++static int template_i2s_startup(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when a PCM substream is closed. Private data can be
++ * freed here.
++ */
++static int template_i2s_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the PCM substream is prepared, can set format, sample
++ * rate, etc.  This function is non atomic and can be called multiple times,
++ * it can refer to the runtime info.
++ */
++static int template_i2s_prepare(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the hardware params are set by application. This
++ * function can also be called multiple times and can allocate buffers
++ * (using snd_pcm_lib_* ). It's non-atomic.
++ */
++static int template_i2s_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++}
++
++/*
++ * Free's resources allocated by hw_params, can be called multiple times
++ */
++static int template_i2s_hw_free(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Starts (Triggers) audio playback or capture.
++ * Usually only needed for DMA
++ */
++static int template_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++}
++
++
++struct snd_soc_cpu_dai template_i2s_dai = {
++	.name = "template-i2s",
++	.id = 0,
++	.type = SND_SOC_DAI_I2S,
++	.suspend = template_i2s_suspend,
++	.resume = template_i2s_resume,
++	.config_sysclk = template_i2s_config_sysclk,
++	.playback = {
++		.channels_min = 2,
++		.channels_max = 2,},
++	.capture = {
++		.channels_min = 2,
++		.channels_max = 2,},
++	.ops = {
++		.startup = template_i2s_startup,
++		.shutdown = template_i2s_shutdown,
++		.prepare = template_i2s_prepare,
++		.trigger = template_i2s_trigger,
++		.hw_params = template_i2s_hw_params,
++		.hw_free = template_i2s_hw_free,},
++	.caps = {
++		.num_modes = ARRAY_SIZE(template_i2s_modes),
++		.mode = template_i2s_modes,},
++};
++
++EXPORT_SYMBOL_GPL(template_i2s_dai);
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,166 @@
++/*
++ * 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 "template-pcm.h"
++
++/* PCM hardware DMA capabilities - platform specific */
++static const struct snd_pcm_hardware template_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,
++};
++
++/*
++ * Called by ALSA when the hardware params are set by application. This
++ * function can also be called multiple times and can allocate buffers
++ * (using snd_pcm_lib_* ). It's non-atomic.
++ */
++static int template_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++}
++
++/*
++ * Free's resources allocated by hw_params, can be called multiple times
++ */
++static int template_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the PCM substream is prepared, can set format, sample
++ * rate, etc.  This function is non atomic and can be called multiple times,
++ * it can refer to the runtime info.
++ */
++static int template_pcm_prepare(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Starts (Triggers) audio playback or capture.
++ * Usually only needed for DMA
++ */
++static int template_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct template_runtime_data *prtd = substream->runtime->private_data;
++	int ret = 0;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		break;
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		break;
++	case SNDRV_PCM_TRIGGER_RESUME:
++		break;
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		break;
++	default:
++		ret = -EINVAL;
++	}
++
++	return ret;
++}
++
++/*
++ * Returns the DMA audio frame position
++ */
++static snd_pcm_uframes_t
++template_pcm_pointer(struct snd_pcm_substream *substream)
++{
++}
++
++ /*
++ * Called by ALSA when a PCM substream is opened, private data can be allocated.
++ */
++static int template_pcm_open(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when a PCM substream is closed. Private data can be
++ * freed here.
++ */
++static int template_pcm_close(struct snd_pcm_substream *substream)
++{
++}
++
++/* map DMA audio buffer into user space */
++static int template_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);
++}
++
++/* ALSA PCM operations */
++struct snd_pcm_ops template_pcm_ops = {
++	.open		= template_pcm_open,
++	.close		= template_pcm_close,
++	.ioctl		= snd_pcm_lib_ioctl,
++	.hw_params	= template_pcm_hw_params,
++	.hw_free	= template_pcm_hw_free,
++	.prepare	= template_pcm_prepare,
++	.trigger	= template_pcm_trigger,
++	.pointer	= template_pcm_pointer,
++	.mmap		= template_pcm_mmap,
++};
++
++/*
++ * Called by ASoC core to free platform DMA.
++ */
++static void template_pcm_free_dma_buffers(struct snd_pcm *pcm)
++{
++}
++
++/*
++ * Called by the ASoC core to create and initialise the platform DMA.
++ */
++int template_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
++	struct snd_pcm *pcm)
++{
++}
++
++/* template audio platform */
++struct snd_soc_platform template_soc_platform = {
++	.name		= "template-audio",
++	.pcm_ops 	= &template_pcm_ops,
++	.pcm_new	= template_pcm_new,
++	.pcm_free	= template_pcm_free_dma_buffers,
++};
++EXPORT_SYMBOL_GPL(template_soc_platform);
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,19 @@
++/*
++ * 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 _TEMPLATE_PCM_H
++#define _TEMPLATE_PCM_H
++
++/* platform specific structs can be declared here */
++
++extern struct snd_soc_cpu_dai template_ac97_dai[3];
++extern struct snd_soc_cpu_dai template_i2s_dai;
++
++/* template platform data */
++extern struct snd_soc_platform template_soc_platform;
++extern struct snd_ac97_bus_ops tempalte_ac97_ops;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,21 @@
++/*
++ * 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 _TEMPLATE_H
++#define _TEMPLATE_H
++
++/* TEMPLATE register space */
++
++#define TEMPLATE_CACHEREGNUM 	10
++
++struct template_codec_setup_data {
++	unsigned short i2c_address;
++};
++
++extern struct snd_soc_codec_dai template_codec_dai;
++extern struct snd_soc_codec_device soc_codec_dev_template_codec;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-machine.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-machine.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,161 @@
++/*
++ *  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 "../codecs/template-codec.h"
++#include "template-pcm.h"
++
++/*
++ * Alsa operations
++ * Only implement the required operations for your platform.
++ * These operations are specific to the machine only.
++ */
++
++ /*
++ * Called by ALSA when a PCM substream is opened, private data can be allocated.
++ */
++static int template_machine_startup(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when a PCM substream is closed. Private data can be
++ * freed here.
++ */
++static int template_machine_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++/*
++ * Called by ALSA when the hardware params are set by application. This
++ * function can also be called multiple times and can allocate buffers
++ * (using snd_pcm_lib_* ). It's non-atomic.
++ */
++static int template_machine_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++}
++
++/*
++ * Free's resources allocated by hw_params, can be called multiple times
++ */
++static int template_machine_hw_free(struct snd_pcm_substream *substream)
++{
++}
++
++/* machine Alsa PCM operations */
++static struct snd_soc_ops template_ops = {
++	.startup = template_machine_startup,
++	.shutdown = template_machine_shutdown,
++	.hw_free = template_machine_hw_free,
++	.hw_params = template_machine_hw_params,
++};
++
++/* machine audio map (connections to the codec pins) */
++static const char *audio_map[][3] = {
++
++	{NULL, NULL, NULL},
++};
++
++/*
++ * Initialise the machine audio subsystem.
++ */
++static int template_machine_init(struct snd_soc_codec *codec)
++{
++	/* mark unused codec pins as NC */
++
++	/* Add template specific controls */
++
++	/* Add template specific widgets */
++
++	/* Set up template specific audio path audio_map */
++
++	/* synchronise subsystem */
++	snd_soc_dapm_sync_endpoints(codec);
++	return 0;
++}
++
++/*
++ * Configure the clocking within the audio subsystem
++ */
++static unsigned int template_config_sysclk(struct snd_soc_pcm_runtime *rtd,
++	struct snd_soc_clock_info *info)
++{
++}
++
++/* template digital audio interface glue - connects codec <--> CPU */
++static struct snd_soc_dai_link template_dai = {
++	.name = "Codec",
++	.stream_name = "Stream Name",
++	.cpu_dai = &template_i2s_dai,
++	.codec_dai = &template_codec_dai,
++	.init = template_machine_init,
++	.config_sysclk = template_config_sysclk,
++};
++
++/* template audio machine driver */
++static struct snd_soc_machine snd_soc_machine_template = {
++	.name = "Machine",
++	.dai_link = &template_dai,
++	.num_links = ARRAY_SIZE(template_dai),
++	.ops = &template_ops,
++};
++
++/* template audio private data */
++static struct codec_priv_setup_data template_codec_setup = {
++	.i2c_address = 0x1b,
++};
++
++/* template audio subsystem */
++static struct snd_soc_device template_snd_devdata = {
++	.machine = &snd_soc_machine_template,
++	.platform = &template_soc_platform,
++	.codec_dev = &soc_codec_dev_wm8731,
++	.codec_data = &template_codec_setup,
++};
++
++static struct platform_device *template_snd_device;
++
++static int __init template_init(void)
++{
++	int ret;
++
++	template_snd_device = platform_device_alloc("soc-audio", -1);
++	if (!template_snd_device)
++		return -ENOMEM;
++
++	platform_set_drvdata(template_snd_device, &template_snd_devdata);
++	template_snd_devdata.dev = &template_snd_device->dev;
++	ret = platform_device_add(template_snd_device);
++
++	if (ret)
++		platform_device_put(template_snd_device);
++
++	return ret;
++}
++
++static void __exit template_exit(void)
++{
++	platform_device_unregister(template_snd_device);
++}
++
++module_init(template_init);
++module_exit(template_exit);
++
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,711 @@
++/*
++ * at91-i2s.c  --  ALSA SoC I2S Audio Layer Platform driver
++ *
++ * Author: Frank Mandarino <fmandarino at endrelia.com>
++ *         Endrelia Technologies Inc.
++ *
++ * Based on pxa2xx Platform drivers by
++ * Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/interrupt.h>
++#include <linux/device.h>
++#include <linux/delay.h>
++#include <linux/clk.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#include <asm/arch/hardware.h>
++#include <asm/arch/at91_pmc.h>
++#include <asm/arch/at91_ssc.h>
++#include <asm/arch/at91_pdc.h>
++
++#include "at91-pcm.h"
++#include "at91-i2s.h"
++
++#if 0
++#define	DBG(x...)	printk(KERN_DEBUG "at91-i2s:" x)
++#else
++#define	DBG(x...)
++#endif
++
++#if defined(CONFIG_ARCH_AT91SAM9260)
++#define NUM_SSC_DEVICES		1
++#else
++#define NUM_SSC_DEVICES		3
++#endif
++
++
++/*
++ * SSC PDC registers required by the PCM DMA engine.
++ */
++static struct at91_pdc_regs pdc_tx_reg = {
++	.xpr		= AT91_PDC_TPR,
++	.xcr		= AT91_PDC_TCR,
++	.xnpr		= AT91_PDC_TNPR,
++	.xncr		= AT91_PDC_TNCR,
++};
++
++static struct at91_pdc_regs pdc_rx_reg = {
++	.xpr		= AT91_PDC_RPR,
++	.xcr		= AT91_PDC_RCR,
++	.xnpr		= AT91_PDC_RNPR,
++	.xncr		= AT91_PDC_RNCR,
++};
++
++/*
++ * SSC & PDC status bits for transmit and receive.
++ */
++static struct at91_ssc_mask ssc_tx_mask = {
++	.ssc_enable	= AT91_SSC_TXEN,
++	.ssc_disable	= AT91_SSC_TXDIS,
++	.ssc_endx	= AT91_SSC_ENDTX,
++	.ssc_endbuf	= AT91_SSC_TXBUFE,
++	.pdc_enable	= AT91_PDC_TXTEN,
++	.pdc_disable	= AT91_PDC_TXTDIS,
++};
++
++static struct at91_ssc_mask ssc_rx_mask = {
++	.ssc_enable	= AT91_SSC_RXEN,
++	.ssc_disable	= AT91_SSC_RXDIS,
++	.ssc_endx	= AT91_SSC_ENDRX,
++	.ssc_endbuf	= AT91_SSC_RXBUFF,
++	.pdc_enable	= AT91_PDC_RXTEN,
++	.pdc_disable	= AT91_PDC_RXTDIS,
++};
++
++
++/*
++ * DMA parameters.
++ */
++static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
++	{{
++	.name		= "SSC0/I2S PCM Stereo out",
++	.pdc		= &pdc_tx_reg,
++	.mask		= &ssc_tx_mask,
++	},
++	{
++	.name		= "SSC0/I2S PCM Stereo in",
++	.pdc		= &pdc_rx_reg,
++	.mask		= &ssc_rx_mask,
++	}},
++#if NUM_SSC_DEVICES == 3
++	{{
++	.name		= "SSC1/I2S PCM Stereo out",
++	.pdc		= &pdc_tx_reg,
++	.mask		= &ssc_tx_mask,
++	},
++	{
++	.name		= "SSC1/I2S PCM Stereo in",
++	.pdc		= &pdc_rx_reg,
++	.mask		= &ssc_rx_mask,
++	}},
++	{{
++	.name		= "SSC2/I2S PCM Stereo out",
++	.pdc		= &pdc_tx_reg,
++	.mask		= &ssc_tx_mask,
++	},
++	{
++	.name		= "SSC1/I2S PCM Stereo in",
++	.pdc		= &pdc_rx_reg,
++	.mask		= &ssc_rx_mask,
++	}},
++#endif
++};
++
++struct at91_ssc_state {
++	u32	ssc_cmr;
++	u32	ssc_rcmr;
++	u32	ssc_rfmr;
++	u32	ssc_tcmr;
++	u32	ssc_tfmr;
++	u32	ssc_sr;
++	u32	ssc_imr;
++};
++
++static struct at91_ssc_info {
++	char		*name;
++	struct at91_ssc_periph ssc;
++	spinlock_t 	lock;		/* lock for dir_mask */
++	unsigned short	dir_mask;	/* 0=unused, 1=playback, 2=capture */
++	unsigned short	initialized;	/* 1=SSC has been initialized */
++	unsigned short	daifmt;
++	unsigned short	cmr_div;
++	unsigned short	tcmr_period;
++	unsigned short	rcmr_period;
++	struct at91_pcm_dma_params *dma_params[2];
++	struct at91_ssc_state ssc_state;
++
++} ssc_info[NUM_SSC_DEVICES] = {
++	{
++	.name		= "ssc0",
++	.lock		= SPIN_LOCK_UNLOCKED,
++	.dir_mask	= 0,
++	.initialized	= 0,
++	},
++#if NUM_SSC_DEVICES == 3
++	{
++	.name		= "ssc1",
++	.lock		= SPIN_LOCK_UNLOCKED,
++	.dir_mask	= 0,
++	.initialized	= 0,
++	},
++	{
++	.name		= "ssc2",
++	.lock		= SPIN_LOCK_UNLOCKED,
++	.dir_mask	= 0,
++	.initialized	= 0,
++	},
++#endif
++};
++
++static unsigned int at91_i2s_sysclk;
++
++/*
++ * SSC interrupt handler.  Passes PDC interrupts to the DMA
++ * interrupt handler in the PCM driver.
++ */
++static irqreturn_t at91_i2s_interrupt(int irq, void *dev_id)
++{
++	struct at91_ssc_info *ssc_p = dev_id;
++	struct at91_pcm_dma_params *dma_params;
++	u32 ssc_sr;
++	int i;
++
++	ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)
++			& at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
++
++	/*
++	 * Loop through the substreams attached to this SSC.  If
++	 * a DMA-related interrupt occurred on that substream, call
++	 * the DMA interrupt handler function, if one has been
++	 * registered in the dma_params structure by the PCM driver.
++	 */
++	for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
++		dma_params = ssc_p->dma_params[i];
++
++		if (dma_params != NULL && dma_params->dma_intr_handler != NULL &&
++			(ssc_sr &
++			(dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf)))
++
++			dma_params->dma_intr_handler(ssc_sr, dma_params->substream);
++	}
++
++	return IRQ_HANDLED;
++}
++
++/*
++ * Startup.  Only that one substream allowed in each direction.
++ */
++static int at91_i2s_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
++	int dir_mask;
++
++	DBG("i2s_startup: SSC_SR=0x%08lx\n",
++			at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
++	dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2;
++
++	spin_lock_irq(&ssc_p->lock);
++	if (ssc_p->dir_mask & dir_mask) {
++		spin_unlock_irq(&ssc_p->lock);
++		return -EBUSY;
++	}
++	ssc_p->dir_mask |= dir_mask;
++	spin_unlock_irq(&ssc_p->lock);
++
++	return 0;
++}
++
++/*
++ * Shutdown.  Clear DMA parameters and shutdown the SSC if there
++ * are no other substreams open.
++ */
++static void at91_i2s_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
++	struct at91_pcm_dma_params *dma_params;
++	int dir, dir_mask;
++
++	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
++	dma_params = ssc_p->dma_params[dir];
++
++	if (dma_params != NULL) {
++		at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
++				dma_params->mask->ssc_disable);
++		DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"),
++			at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
++
++		dma_params->ssc_base = NULL;
++		dma_params->substream = NULL;
++		ssc_p->dma_params[dir] = NULL;
++	}
++
++	dir_mask = 1 << dir;
++
++	spin_lock_irq(&ssc_p->lock);
++	ssc_p->dir_mask &= ~dir_mask;
++	if (!ssc_p->dir_mask) {
++		/* Shutdown the SSC clock. */
++		DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
++		at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
++
++		if (ssc_p->initialized) {
++			free_irq(ssc_p->ssc.pid, ssc_p);
++			ssc_p->initialized = 0;
++		}
++
++		/* Reset the SSC */
++		at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
++
++		/* Clear the SSC dividers */
++		ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
++	}
++	spin_unlock_irq(&ssc_p->lock);
++}
++
++/*
++ * Record the SSC system clock rate.
++ */
++static int at91_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
++		int clk_id, unsigned int freq, int dir)
++{
++	/*
++	 * The only clock supplied to the SSC is the AT91 master clock,
++	 * which is only used if the SSC is generating BCLK and/or
++	 * LRC clocks.
++	 */
++	switch (clk_id) {
++	case AT91_SYSCLK_MCK:
++		at91_i2s_sysclk = freq;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++/*
++ * Record the DAI format for use in hw_params().
++ */
++static int at91_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int fmt)
++{
++	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
++
++	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S)
++		return -EINVAL;
++
++	ssc_p->daifmt = fmt;
++	return 0;
++}
++
++/*
++ * Record SSC clock dividers for use in hw_params().
++ */
++static int at91_i2s_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
++	int div_id, int div)
++{
++	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
++
++	switch (div_id) {
++	case AT91SSC_CMR_DIV:
++		/*
++		 * The same master clock divider is used for both
++		 * transmit and receive, so if a value has already
++		 * been set, it must match this value.
++		 */
++		if (ssc_p->cmr_div == 0)
++			ssc_p->cmr_div = div;
++		else
++			if (div != ssc_p->cmr_div)
++				return -EBUSY;
++		break;
++
++	case AT91SSC_TCMR_PERIOD:
++		ssc_p->tcmr_period = div;
++		break;
++
++	case AT91SSC_RCMR_PERIOD:
++		ssc_p->rcmr_period = div;
++		break;
++
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++/*
++ * Configure the SSC.
++ */
++static int at91_i2s_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	int id = rtd->dai->cpu_dai->id;
++	struct at91_ssc_info *ssc_p = &ssc_info[id];
++	struct at91_pcm_dma_params *dma_params;
++	int dir, channels, bits;
++	u32 tfmr, rfmr, tcmr, rcmr;
++	int ret;
++
++	/*
++	 * Currently, there is only one set of dma params for
++	 * each direction.  If more are added, this code will
++	 * have to be changed to select the proper set.
++	 */
++	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
++
++	dma_params = &ssc_dma_params[id][dir];
++	dma_params->ssc_base = ssc_p->ssc.base;
++	dma_params->substream = substream;
++
++	ssc_p->dma_params[dir] = dma_params;
++
++	/*
++	 * The cpu_dai->dma_data field is only used to communicate the
++	 * appropriate DMA parameters to the pcm driver hw_params()
++	 * function.  It should not be used for other purposes
++	 * as it is common to all substreams.
++	 */
++	rtd->dai->cpu_dai->dma_data = dma_params;
++
++	channels = params_channels(params);
++
++	/*
++	 * The SSC only supports up to 16-bit samples in I2S format, due
++	 * to the size of the Frame Mode Register FSLEN field.  Also, I2S
++	 * implies signed data.
++	 */
++	bits = 16;
++	dma_params->pdc_xfer_size = 2;
++
++	/*
++	 * Compute SSC register settings.
++	 */
++	switch (ssc_p->daifmt) {
++	case SND_SOC_DAIFMT_CBS_CFS:
++		/*
++		 * SSC provides BCLK and LRC clocks.
++		 *
++		 * The SSC transmit clock is generated from the MCK divider
++		 * and used to generate the SSC receive clock and the BCLK
++		 * output on the SSC TK line.
++		 */
++		rcmr =	  (( ssc_p->rcmr_period		<< 24) & AT91_SSC_PERIOD)
++			| (( 1				<< 16) & AT91_SSC_STTDLY)
++			| (( AT91_SSC_START_TX_RX	     ) & AT91_SSC_START)
++			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
++			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
++			| (( AT91_SSC_CKS_CLOCK		     ) & AT91_SSC_CKS);
++
++		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
++			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
++			| (( 0				<< 16) & AT91_SSC_FSLEN)
++			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
++			| (( 1				<<  7) & AT91_SSC_MSBF)
++			| (( 0				<<  5) & AT91_SSC_LOOP)
++			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
++
++		tcmr =	  (( ssc_p->tcmr_period		<< 24) & AT91_SSC_PERIOD)
++			| (( 1				<< 16) & AT91_SSC_STTDLY)
++			| (( AT91_SSC_START_FALLING_RF       ) & AT91_SSC_START)
++			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
++			| (( AT91_SSC_CKO_CONTINUOUS	     ) & AT91_SSC_CKO)
++			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
++
++		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
++			| (( 0				<< 23) & AT91_SSC_FSDEN)
++			| (( AT91_SSC_FSOS_NEGATIVE	     ) & AT91_SSC_FSOS)
++			| (((bits - 1)			<< 16) & AT91_SSC_FSLEN)
++			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
++			| (( 1				<<  7) & AT91_SSC_MSBF)
++			| (( 0				<<  5) & AT91_SSC_DATDEF)
++			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
++		break;
++
++	case SND_SOC_DAIFMT_CBM_CFM:
++
++#ifdef EXAMPLE_UNTESTED_CODEC_MASTER_CONFIGURATION
++		/*
++		 * CODEC supplies BCLK and LRC clocks.
++		 *
++		 * Assumes BCLK is supplied on SSC RK line.
++		 */
++		rcmr =	  (( 0				<< 24) & AT91_SSC_PERIOD)
++			| (( 1				<< 16) & AT91_SSC_STTDLY)
++			| (( AT91_SSC_START_FALLING_RF	     ) & AT91_SSC_START)
++			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
++			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
++			| (( AT91_SSC_CKS_PIN		     ) & AT91_SSC_CKS);
++
++		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
++			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
++			| (( 0				<< 16) & AT91_SSC_FSLEN)
++			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
++			| (( 1				<<  7) & AT91_SSC_MSBF)
++			| (( 0				<<  5) & AT91_SSC_LOOP)
++			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
++
++		tcmr =	  (( 0				<< 24) & AT91_SSC_PERIOD)
++			| (( 1				<< 16) & AT91_SSC_STTDLY)
++			| (( AT91_SSC_START_FALLING_RF       ) & AT91_SSC_START)
++			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
++			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
++			| (( AT91_SSC_CKS_CLOCK		     ) & AT91_SSC_CKS);
++
++		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
++			| (( 0				<< 23) & AT91_SSC_FSDEN)
++			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
++			| (( 0				<< 16) & AT91_SSC_FSLEN)
++			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
++			| (( 1				<<  7) & AT91_SSC_MSBF)
++			| (( 0				<<  5) & AT91_SSC_DATDEF)
++			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
++		break;
++#endif
++	case SND_SOC_DAIFMT_CBS_CFM:
++	case SND_SOC_DAIFMT_CBM_CFS:
++	default:
++		printk(KERN_WARNING "at91-i2s: unsupported DAI format 0x%x.\n",
++			ssc_p->daifmt);
++		return -EINVAL;
++		break;
++	}
++	DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr);
++
++	if (!ssc_p->initialized) {
++
++		/* Enable PMC peripheral clock for this SSC */
++		DBG("Starting pid %d clock\n", ssc_p->ssc.pid);
++		at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
++
++		/* Reset the SSC and its PDC registers */
++		at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
++
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RPR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RCR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RNPR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RNCR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TPR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TCR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TNPR, 0);
++		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TNCR, 0);
++
++		if ((ret = request_irq(ssc_p->ssc.pid, at91_i2s_interrupt,
++					0, ssc_p->name, ssc_p)) < 0) {
++			printk(KERN_WARNING "at91-i2s: request_irq failure\n");
++
++			DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
++			at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
++			return ret;
++		}
++
++		ssc_p->initialized = 1;
++	}
++
++	/* set SSC clock mode register */
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div);
++
++	/* set receive clock mode and format */
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr);
++
++	/* set transmit clock mode and format */
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr);
++
++	DBG("hw_params: SSC initialized\n");
++	return 0;
++}
++
++
++static int at91_i2s_prepare(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
++	struct at91_pcm_dma_params *dma_params;
++	int dir;
++
++	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
++	dma_params = ssc_p->dma_params[dir];
++
++	at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
++			dma_params->mask->ssc_enable);
++
++	DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "transmit" : "receive",
++		at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR));
++	return 0;
++}
++
++
++#ifdef CONFIG_PM
++static int at91_i2s_suspend(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *cpu_dai)
++{
++	struct at91_ssc_info *ssc_p;
++
++	if(!cpu_dai->active)
++		return 0;
++
++	ssc_p = &ssc_info[cpu_dai->id];
++
++	/* Save the status register before disabling transmit and receive. */
++	ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
++			AT91_SSC_TXDIS | AT91_SSC_RXDIS);
++
++	/* Save the current interrupt mask, then disable unmasked interrupts. */
++	ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr);
++
++	ssc_p->ssc_state.ssc_cmr  = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR);
++	ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR);
++	ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR);
++	ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR);
++	ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR);
++
++	return 0;
++}
++
++static int at91_i2s_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *cpu_dai)
++{
++	struct at91_ssc_info *ssc_p;
++
++	if(!cpu_dai->active)
++		return 0;
++
++	ssc_p = &ssc_info[cpu_dai->id];
++
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr);
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR,  ssc_p->ssc_state.ssc_cmr);
++
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER,  ssc_p->ssc_state.ssc_imr);
++
++	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
++		((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) |
++		((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0));
++
++	return 0;
++}
++
++#else
++#define at91_i2s_suspend	NULL
++#define at91_i2s_resume		NULL
++#endif
++
++#define AT91_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)
++
++struct snd_soc_cpu_dai at91_i2s_dai[NUM_SSC_DEVICES] = {
++	{	.name = "at91_ssc0/i2s",
++		.id = 0,
++		.type = SND_SOC_DAI_I2S,
++		.suspend = at91_i2s_suspend,
++		.resume = at91_i2s_resume,
++		.playback = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = AT91_I2S_RATES,
++			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++		.capture = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = AT91_I2S_RATES,
++			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++		.ops = {
++			.startup = at91_i2s_startup,
++			.shutdown = at91_i2s_shutdown,
++			.prepare = at91_i2s_prepare,
++			.hw_params = at91_i2s_hw_params,},
++		.dai_ops = {
++			.set_sysclk = at91_i2s_set_dai_sysclk,
++			.set_fmt = at91_i2s_set_dai_fmt,
++			.set_clkdiv = at91_i2s_set_dai_clkdiv,},
++		.private_data = &ssc_info[0].ssc,
++	},
++#if NUM_SSC_DEVICES == 3
++	{	.name = "at91_ssc1/i2s",
++		.id = 1,
++		.type = SND_SOC_DAI_I2S,
++		.suspend = at91_i2s_suspend,
++		.resume = at91_i2s_resume,
++		.playback = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = AT91_I2S_RATES,
++			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++		.capture = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = AT91_I2S_RATES,
++			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++		.ops = {
++			.startup = at91_i2s_startup,
++			.shutdown = at91_i2s_shutdown,
++			.prepare = at91_i2s_prepare,
++			.hw_params = at91_i2s_hw_params,},
++		.dai_ops = {
++			.set_sysclk = at91_i2s_set_dai_sysclk,
++			.set_fmt = at91_i2s_set_dai_fmt,
++			.set_clkdiv = at91_i2s_set_dai_clkdiv,},
++		.private_data = &ssc_info[1].ssc,
++	},
++	{	.name = "at91_ssc2/i2s",
++		.id = 2,
++		.type = SND_SOC_DAI_I2S,
++		.suspend = at91_i2s_suspend,
++		.resume = at91_i2s_resume,
++		.playback = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = AT91_I2S_RATES,
++			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++		.capture = {
++			.channels_min = 1,
++			.channels_max = 2,
++			.rates = AT91_I2S_RATES,
++			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++		.ops = {
++			.startup = at91_i2s_startup,
++			.shutdown = at91_i2s_shutdown,
++			.prepare = at91_i2s_prepare,
++			.hw_params = at91_i2s_hw_params,},
++		.dai_ops = {
++			.set_sysclk = at91_i2s_set_dai_sysclk,
++			.set_fmt = at91_i2s_set_dai_fmt,
++			.set_clkdiv = at91_i2s_set_dai_clkdiv,},
++		.private_data = &ssc_info[2].ssc,
++	},
++#endif
++};
++
++EXPORT_SYMBOL_GPL(at91_i2s_dai);
++
++/* Module information */
++MODULE_AUTHOR("Frank Mandarino, fmandarino at endrelia.com, www.endrelia.com");
++MODULE_DESCRIPTION("AT91 I2S ASoC Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,432 @@
++/*
++ * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
++ *
++ * Author:	Frank Mandarino <fmandarino at endrelia.com>
++ *		Endrelia Technologies Inc.
++ * Created:	Mar 3, 2006
++ *
++ * Based on pxa2xx-pcm.c by:
++ *
++ * 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/arch/hardware.h>
++#include <asm/arch/at91_ssc.h>
++#include <asm/arch/at91_pdc.h>
++
++#include "at91-pcm.h"
++
++#if 0
++#define	DBG(x...)	printk(KERN_INFO "at91-pcm: " x)
++#else
++#define	DBG(x...)
++#endif
++
++static const struct snd_pcm_hardware at91_pcm_hardware = {
++	.info			= SNDRV_PCM_INFO_MMAP |
++				  SNDRV_PCM_INFO_MMAP_VALID |
++				  SNDRV_PCM_INFO_INTERLEAVED |
++				  SNDRV_PCM_INFO_PAUSE,
++	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
++	.period_bytes_min	= 32,
++	.period_bytes_max	= 8192,
++	.periods_min		= 2,
++	.periods_max		= 1024,
++	.buffer_bytes_max	= 32 * 1024,
++};
++
++struct at91_runtime_data {
++	struct at91_pcm_dma_params *params;
++	dma_addr_t dma_buffer;			/* physical address of dma buffer */
++	dma_addr_t dma_buffer_end;		/* first address beyond DMA buffer */
++	size_t period_size;
++	dma_addr_t period_ptr;			/* physical address of next period */
++	u32 pdc_xpr_save;			/* PDC register save */
++	u32 pdc_xcr_save;
++	u32 pdc_xnpr_save;
++	u32 pdc_xncr_save;
++};
++
++static void at91_pcm_dma_irq(u32 ssc_sr,
++	struct snd_pcm_substream *substream)
++{
++	struct at91_runtime_data *prtd = substream->runtime->private_data;
++	struct at91_pcm_dma_params *params = prtd->params;
++	static int count = 0;
++
++	count++;
++
++	if (ssc_sr & params->mask->ssc_endbuf) {
++
++		printk(KERN_WARNING
++			"at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
++			substream->stream == SNDRV_PCM_STREAM_PLAYBACK
++				? "underrun" : "overrun",
++			params->name, ssc_sr, count);
++
++		/* re-start the PDC */
++		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
++
++		prtd->period_ptr += prtd->period_size;
++		if (prtd->period_ptr >= prtd->dma_buffer_end) {
++			prtd->period_ptr = prtd->dma_buffer;
++		}
++
++		at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
++		at91_ssc_write(params->ssc_base + params->pdc->xcr,
++				prtd->period_size / params->pdc_xfer_size);
++
++		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
++	}
++
++	if (ssc_sr & params->mask->ssc_endx) {
++
++		/* Load the PDC next pointer and counter registers */
++		prtd->period_ptr += prtd->period_size;
++		if (prtd->period_ptr >= prtd->dma_buffer_end) {
++			prtd->period_ptr = prtd->dma_buffer;
++		}
++		at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
++		at91_ssc_write(params->ssc_base + params->pdc->xncr,
++				prtd->period_size / params->pdc_xfer_size);
++	}
++
++	snd_pcm_period_elapsed(substream);
++}
++
++static int at91_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct at91_runtime_data *prtd = runtime->private_data;
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++
++	/* this may get called several times by oss emulation
++	 * with different params */
++
++	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
++	runtime->dma_bytes = params_buffer_bytes(params);
++
++	prtd->params = rtd->dai->cpu_dai->dma_data;
++	prtd->params->dma_intr_handler = at91_pcm_dma_irq;
++
++	prtd->dma_buffer = runtime->dma_addr;
++	prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
++	prtd->period_size = params_period_bytes(params);
++
++	DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n",
++		prtd->params->name, runtime->dma_bytes, prtd->period_size);
++	return 0;
++}
++
++static int at91_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++	struct at91_runtime_data *prtd = substream->runtime->private_data;
++	struct at91_pcm_dma_params *params = prtd->params;
++
++	if (params != NULL) {
++		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
++		prtd->params->dma_intr_handler = NULL;
++	}
++
++	return 0;
++}
++
++static int at91_pcm_prepare(struct snd_pcm_substream *substream)
++{
++	struct at91_runtime_data *prtd = substream->runtime->private_data;
++	struct at91_pcm_dma_params *params = prtd->params;
++
++	at91_ssc_write(params->ssc_base + AT91_SSC_IDR,
++			params->mask->ssc_endx | params->mask->ssc_endbuf);
++
++	at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
++	return 0;
++}
++
++static int at91_pcm_trigger(struct snd_pcm_substream *substream,
++	int cmd)
++{
++	struct at91_runtime_data *prtd = substream->runtime->private_data;
++	struct at91_pcm_dma_params *params = prtd->params;
++	int ret = 0;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		prtd->period_ptr = prtd->dma_buffer;
++
++		at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
++		at91_ssc_write(params->ssc_base + params->pdc->xcr,
++				prtd->period_size / params->pdc_xfer_size);
++
++		prtd->period_ptr += prtd->period_size;
++		at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
++		at91_ssc_write(params->ssc_base + params->pdc->xncr,
++				prtd->period_size / params->pdc_xfer_size);
++
++		DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n",
++			(unsigned long) prtd->period_ptr,
++			at91_ssc_read(params->ssc_base + params->pdc->xpr),
++			at91_ssc_read(params->ssc_base + params->pdc->xcr),
++			at91_ssc_read(params->ssc_base + params->pdc->xnpr),
++			at91_ssc_read(params->ssc_base + params->pdc->xncr));
++
++		at91_ssc_write(params->ssc_base + AT91_SSC_IER,
++			params->mask->ssc_endx | params->mask->ssc_endbuf);
++
++		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
++
++		DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc_base + AT91_SSC_SR),
++					at91_ssc_read(params->ssc_base + AT91_SSC_IER));
++		break;
++
++	case SNDRV_PCM_TRIGGER_STOP:
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
++		break;
++
++	case SNDRV_PCM_TRIGGER_RESUME:
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
++		break;
++
++	default:
++		ret = -EINVAL;
++	}
++
++	return ret;
++}
++
++static snd_pcm_uframes_t at91_pcm_pointer(
++	struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct at91_runtime_data *prtd = runtime->private_data;
++	struct at91_pcm_dma_params *params = prtd->params;
++	dma_addr_t ptr;
++	snd_pcm_uframes_t x;
++
++	ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr);
++	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
++
++	if (x == runtime->buffer_size)
++		x = 0;
++	return x;
++}
++
++static int at91_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct at91_runtime_data *prtd;
++	int ret = 0;
++
++	snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware);
++
++	/* ensure that buffer size is a multiple of period size */
++	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
++	if (ret < 0)
++		goto out;
++
++	prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL);
++	if (prtd == NULL) {
++		ret = -ENOMEM;
++		goto out;
++	}
++	runtime->private_data = prtd;
++
++ out:
++	return ret;
++}
++
++static int at91_pcm_close(struct snd_pcm_substream *substream)
++{
++	struct at91_runtime_data *prtd = substream->runtime->private_data;
++
++	kfree(prtd);
++	return 0;
++}
++
++static int at91_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 at91_pcm_ops = {
++	.open		= at91_pcm_open,
++	.close		= at91_pcm_close,
++	.ioctl		= snd_pcm_lib_ioctl,
++	.hw_params	= at91_pcm_hw_params,
++	.hw_free	= at91_pcm_hw_free,
++	.prepare	= at91_pcm_prepare,
++	.trigger	= at91_pcm_trigger,
++	.pointer	= at91_pcm_pointer,
++	.mmap		= at91_pcm_mmap,
++};
++
++static int at91_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 = at91_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);
++
++	DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
++		(void *) buf->area,
++		(void *) buf->addr,
++		size);
++
++	if (!buf->area)
++		return -ENOMEM;
++
++	buf->bytes = size;
++	return 0;
++}
++
++static u64 at91_pcm_dmamask = 0xffffffff;
++
++static int at91_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 = &at91_pcm_dmamask;
++	if (!card->dev->coherent_dma_mask)
++		card->dev->coherent_dma_mask = 0xffffffff;
++
++	if (dai->playback.channels_min) {
++		ret = at91_pcm_preallocate_dma_buffer(pcm,
++			SNDRV_PCM_STREAM_PLAYBACK);
++		if (ret)
++			goto out;
++	}
++
++	if (dai->capture.channels_min) {
++		ret = at91_pcm_preallocate_dma_buffer(pcm,
++			SNDRV_PCM_STREAM_CAPTURE);
++		if (ret)
++			goto out;
++	}
++ out:
++	return ret;
++}
++
++static void at91_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;
++	}
++}
++
++#ifdef CONFIG_PM
++static int at91_pcm_suspend(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++	struct snd_pcm_runtime *runtime = dai->runtime;
++	struct at91_runtime_data *prtd;
++	struct at91_pcm_dma_params *params;
++
++	if (!runtime)
++		return 0;
++
++	prtd = runtime->private_data;
++	params = prtd->params;
++
++	/* disable the PDC and save the PDC registers */
++
++	at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
++
++	prtd->pdc_xpr_save  = at91_ssc_read(params->ssc_base + params->pdc->xpr);
++	prtd->pdc_xcr_save  = at91_ssc_read(params->ssc_base + params->pdc->xcr);
++	prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr);
++	prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr);
++
++	return 0;
++}
++
++static int at91_pcm_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++	struct snd_pcm_runtime *runtime = dai->runtime;
++	struct at91_runtime_data *prtd;
++	struct at91_pcm_dma_params *params;
++
++	if (!runtime)
++		return 0;
++
++	prtd = runtime->private_data;
++	params = prtd->params;
++
++	/* restore the PDC registers and enable the PDC */
++	at91_ssc_write(params->ssc_base + params->pdc->xpr,  prtd->pdc_xpr_save);
++	at91_ssc_write(params->ssc_base + params->pdc->xcr,  prtd->pdc_xcr_save);
++	at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save);
++	at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save);
++
++	at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
++	return 0;
++}
++#else
++#define at91_pcm_suspend	NULL
++#define at91_pcm_resume		NULL
++#endif
++
++struct snd_soc_platform at91_soc_platform = {
++	.name		= "at91-audio",
++	.pcm_ops 	= &at91_pcm_ops,
++	.pcm_new	= at91_pcm_new,
++	.pcm_free	= at91_pcm_free_dma_buffers,
++	.suspend	= at91_pcm_suspend,
++	.resume		= at91_pcm_resume,
++};
++
++EXPORT_SYMBOL_GPL(at91_soc_platform);
++
++MODULE_AUTHOR("Frank Mandarino <fmandarino at endrelia.com>");
++MODULE_DESCRIPTION("Atmel AT91 PCM module");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,72 @@
++/*
++ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
++ *
++ * Author:	Frank Mandarino <fmandarino at endrelia.com>
++ *		Endrelia Technologies Inc.
++ * Created:	Mar 3, 2006
++ *
++ * Based on pxa2xx-pcm.h by:
++ *
++ * 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 _AT91_PCM_H
++#define _AT91_PCM_H
++
++#include <asm/arch/hardware.h>
++
++struct at91_ssc_periph {
++	void __iomem	*base;
++	u32		pid;
++};
++
++/*
++ * Registers and status bits that are required by the PCM driver.
++ */
++struct at91_pdc_regs {
++	unsigned int	xpr;		/* PDC recv/trans pointer */
++	unsigned int	xcr;		/* PDC recv/trans counter */
++	unsigned int	xnpr;		/* PDC next recv/trans pointer */
++	unsigned int	xncr;		/* PDC next recv/trans counter */
++	unsigned int	ptcr;		/* PDC transfer control */
++};
++
++struct at91_ssc_mask {
++	u32	ssc_enable;		/* SSC recv/trans enable */
++	u32	ssc_disable;		/* SSC recv/trans disable */
++	u32	ssc_endx;		/* SSC ENDTX or ENDRX */
++	u32	ssc_endbuf;		/* SSC TXBUFE or RXBUFF */
++	u32	pdc_enable;		/* PDC recv/trans enable */
++	u32	pdc_disable;		/* PDC recv/trans disable */
++};
++
++/*
++ * This structure, shared between the PCM driver and the interface,
++ * contains all information required by the PCM driver to perform the
++ * PDC DMA operation.  All fields except dma_intr_handler() are initialized
++ * by the interface.  The dms_intr_handler() pointer is set by the PCM
++ * driver and called by the interface SSC interrupt handler if it is
++ * non-NULL.
++ */
++struct at91_pcm_dma_params {
++	char *name;			/* stream identifier */
++	int pdc_xfer_size;		/* PDC counter increment in bytes */
++	void __iomem *ssc_base;		/* SSC base address */
++	struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
++	struct at91_ssc_mask *mask;/* SSC & PDC status bits */
++	struct snd_pcm_substream *substream;
++	void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
++};
++
++extern struct snd_soc_platform at91_soc_platform;
++
++#define at91_ssc_read(a)	((unsigned long) __raw_readl(a))
++#define at91_ssc_write(a,v)	__raw_writel((v),(a))
++
++#endif /* _AT91_PCM_H */
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,42 @@
++/*
++ * linux/sound/arm/pxa2xx-ssp.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 _PXA2XX_SSP_H
++#define _PXA2XX_SSP_H
++
++/* pxa2xx DAI SSP ID's */
++#define PXA2XX_DAI_SSP1			0
++#define PXA2XX_DAI_SSP2			1
++#define PXA2XX_DAI_SSP3			2
++
++/* SSP clock sources */
++#define PXA2XX_SSP_CLK_PLL	0
++#define PXA2XX_SSP_CLK_EXT	1
++#define PXA2XX_SSP_CLK_NET	2
++#define PXA2XX_SSP_CLK_AUDIO	3
++
++/* SSP audio dividers */
++#define PXA2XX_SSP_AUDIO_DIV_ACDS		0
++#define PXA2XX_SSP_AUDIO_DIV_SCDB		1
++#define PXA2XX_SSP_DIV_SCR				2
++
++/* SSP ACDS audio dividers values */
++#define PXA2XX_SSP_CLK_AUDIO_DIV_1		0
++#define PXA2XX_SSP_CLK_AUDIO_DIV_2		1
++#define PXA2XX_SSP_CLK_AUDIO_DIV_4		2
++#define PXA2XX_SSP_CLK_AUDIO_DIV_8		3
++#define PXA2XX_SSP_CLK_AUDIO_DIV_16	4
++#define PXA2XX_SSP_CLK_AUDIO_DIV_32	5
++
++/* SSP divider bypass */
++#define PXA2XX_SSP_CLK_SCDB_4		0
++#define PXA2XX_SSP_CLK_SCDB_1		1
++
++extern struct snd_soc_cpu_dai pxa_ssp_dai[3];
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,22 @@
++/*
++ * linux/sound/arm/pxa2xx-ac97.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 _PXA2XX_AC97_H
++#define _PXA2XX_AC97_H
++
++/* pxa2xx DAI ID's */
++#define PXA2XX_DAI_AC97_HIFI	0
++#define PXA2XX_DAI_AC97_AUX		1
++#define PXA2XX_DAI_AC97_MIC		2
++
++extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
++
++/* platform data */
++extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,20 @@
++/*
++ * linux/sound/arm/pxa2xx-i2s.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 _PXA2XX_I2S_H
++#define _PXA2XX_I2S_H
++
++/* pxa2xx DAI ID's */
++#define PXA2XX_DAI_I2S			0
++
++/* I2S clock */
++#define PXA2XX_I2S_SYSCLK		0
++
++extern struct snd_soc_cpu_dai pxa_i2s_dai;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Kconfig
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Kconfig	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,30 @@
++menu "SoC Audio for the Samsung S3C24XX"
++
++config SND_S3C24XX_SOC
++	tristate "SoC Audio for the Samsung S3C24XX chips"
++	depends on ARCH_S3C2410 && SND
++	select SND_PCM
++	help
++	  Say Y or M if you want to add support for codecs attached to
++	  the S3C24XX AC97, I2S or SSP interface. You will also need
++	  to select the audio interfaces to support below.
++
++config SND_S3C24XX_SOC_I2S
++	tristate
++
++config SND_S3C24XX_SOC_SMDK2440
++	tristate "SoC I2S Audio support for SMDK2440"
++	depends on SND_S3C24XX_SOC && MACH_SMDK
++	select SND_S3C24XX_SOC_I2S
++	help
++	  Say Y if you want to add support for SoC audio on SMDK2440
++
++config SND_S3C24XX_SOC_SMDK2440_WM8753
++	tristate "SoC I2S Audio support for S3C24XX - WM8753"
++	depends on SND_S3C24XX_SOC && MACH_SMDK
++	select SND_S3C24XX_SOC_I2S
++	help
++	  Say Y if you want to add support for SoC audio on smdk2440
++	  with the WM8753.
++endmenu
++
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Makefile
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Makefile	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,13 @@
++# S3c24XX Platform Support
++snd-soc-s3c24xx-objs := s3c24xx-pcm.o
++snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
++
++obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
++obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
++
++# S3C24XX Machine Support
++snd-soc-smdk2440-objs := smdk2440.o
++snd-soc-smdk2440-wm8753-objs := smdk2440_wm8753.o
++
++obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440) += snd-soc-smdk2440.o
++obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440_WM8753) += snd-soc-smdk2440-wm8753.o
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,484 @@
++/*
++ * s3c24xx-i2s.c  --  ALSA Soc Audio Layer
++ *
++ * (c) 2006 Wolfson Microelectronics PLC.
++ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ * (c) 2004-2005 Simtec Electronics
++ *	http://armlinux.simtec.co.uk/
++ *	Ben Dooks <ben at simtec.co.uk>
++ *
++ *  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
++ *    11th Dec 2006   Merged with Simtec driver
++ *    10th Nov 2006   Initial version.
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/device.h>
++#include <linux/delay.h>
++#include <linux/clk.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#include <asm/hardware.h>
++#include <asm/io.h>
++#include <asm/arch/regs-iis.h>
++#include <asm/arch/regs-gpio.h>
++#include <asm/arch/regs-clock.h>
++#include <asm/arch/audio.h>
++#include <asm/dma.h>
++#include <asm/arch/dma.h>
++
++#include "s3c24xx-pcm.h"
++#include "s3c24xx-i2s.h"
++
++#define S3C24XX_I2S_DEBUG 0
++#if S3C24XX_I2S_DEBUG
++#define DBG(x...) printk(KERN_DEBUG x)
++#else
++#define DBG(x...)
++#endif
++
++static struct s3c2410_dma_client s3c24xx_dma_client_out = {
++	.name = "I2S PCM Stereo out"
++};
++
++static struct s3c2410_dma_client s3c24xx_dma_client_in = {
++	.name = "I2S PCM Stereo in"
++};
++
++static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_out = {
++	.client		= &s3c24xx_dma_client_out,
++	.channel	= DMACH_I2S_OUT,
++	.dma_addr	= S3C2410_PA_IIS + S3C2410_IISFIFO
++};
++
++static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_in = {
++	.client		= &s3c24xx_dma_client_in,
++	.channel	= DMACH_I2S_IN,
++	.dma_addr	= S3C2410_PA_IIS + S3C2410_IISFIFO
++};
++
++struct s3c24xx_i2s_info {
++	void __iomem	*regs;
++	struct clk 		*iis_clk;
++};
++static struct s3c24xx_i2s_info s3c24xx_i2s;
++
++static void s3c24xx_snd_txctrl(int on)
++{
++	unsigned long iisfcon;
++	unsigned long iiscon;
++	unsigned long iismod;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
++	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
++	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++
++	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
++
++	if (on) {
++		iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE;
++		iiscon  |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN;
++		iiscon  &= ~S3C2410_IISCON_TXIDLE;
++		iismod  |= S3C2410_IISMOD_TXMODE;
++
++		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
++		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
++		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
++	} else {
++		/* note, we have to disable the FIFOs otherwise bad things
++		 * seem to happen when the DMA stops. According to the
++		 * Samsung supplied kernel, this should allow the DMA
++		 * engine and FIFOs to reset. If this isn't allowed, the
++		 * DMA engine will simply freeze randomly.
++         */
++
++		iisfcon &= ~S3C2410_IISFCON_TXENABLE;
++		iisfcon &= ~S3C2410_IISFCON_TXDMA;
++		iiscon  |=  S3C2410_IISCON_TXIDLE;
++		iiscon  &= ~S3C2410_IISCON_TXDMAEN;
++		iismod  &= ~S3C2410_IISMOD_TXMODE;
++
++		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
++		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
++		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
++	}
++
++	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
++
++}
++
++static void s3c24xx_snd_rxctrl(int on)
++{
++	unsigned long iisfcon;
++	unsigned long iiscon;
++	unsigned long iismod;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
++	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
++	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++
++	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
++
++	if (on) {
++		iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE;
++		iiscon  |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN;
++		iiscon  &= ~S3C2410_IISCON_RXIDLE;
++		iismod  |= S3C2410_IISMOD_RXMODE;
++
++		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
++		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
++		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
++	} else {
++        /* note, we have to disable the FIFOs otherwise bad things
++         * seem to happen when the DMA stops. According to the
++         * Samsung supplied kernel, this should allow the DMA
++         * engine and FIFOs to reset. If this isn't allowed, the
++         * DMA engine will simply freeze randomly.
++         */
++
++        iisfcon &= ~S3C2410_IISFCON_RXENABLE;
++        iisfcon &= ~S3C2410_IISFCON_RXDMA;
++        iiscon  |= S3C2410_IISCON_RXIDLE;
++        iiscon  &= ~S3C2410_IISCON_RXDMAEN;
++		iismod  &= ~S3C2410_IISMOD_RXMODE;
++
++		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
++		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
++		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
++	}
++
++	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
++
++}
++
++/* s3c24xx_snd_lrsync
++ *
++ * Wait for the LR signal to allow synchronisation to the L/R clock
++ * from the DAC. May only be needed for slave mode.
++*/
++
++static int s3c24xx_snd_lrsync()
++{
++	unsigned long iiscon;
++	int timeout = 10000;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	while (1) {
++		iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
++		if (iiscon & S3C2410_IISCON_LRINDEX)
++			break;
++
++		if (--timeout < 0)
++			return -ETIMEDOUT;
++	}
++
++	return 0;
++}
++
++/* s3c24xx_snd_is_clkmaster()
++ *
++ * Check whether CPU is the master or slave
++*/
++
++static inline int s3c24xx_snd_is_clkmaster()
++{
++	DBG("Entered %s\n", __FUNCTION__);
++
++	return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
++}
++
++static int s3c24xx_i2s_startup(struct snd_pcm_substream *substream)
++{
++	DBG("Entered %s\n", __FUNCTION__);
++
++	return 0;
++}
++
++/*
++ * Set S3C24xx I2S DAI format
++ */
++static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
++		unsigned int fmt)
++{
++	unsigned long iismod;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++	DBG("hw_params r: IISMOD: %lx \n", iismod);
++
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		iismod |= S3C2410_IISMOD_SLAVE;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_LEFT_J:
++		iismod |= S3C2410_IISMOD_MSB;
++		break;
++	case SND_SOC_DAIFMT_I2S:
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
++	DBG("hw_params w: IISMOD: %lx \n", iismod);
++	return 0;
++}
++
++static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	unsigned long iismod;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
++	else
++		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in;
++
++	/* Working copies of register */
++	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++	DBG("hw_params r: IISMOD: %lx\n", iismod);
++
++	switch (params_format(params)) {
++	case SNDRV_PCM_FORMAT_S8:
++		break;
++	case SNDRV_PCM_FORMAT_S16_LE:
++		iismod |= S3C2410_IISMOD_16BIT;
++		break;
++	}
++
++	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
++	DBG("hw_params w: IISMOD: %lx\n", iismod);
++	return 0;
++}
++
++static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	int ret = 0;
++	unsigned long flags;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++	case SNDRV_PCM_TRIGGER_RESUME:
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		if (!s3c24xx_snd_is_clkmaster()) {
++			ret = s3c24xx_snd_lrsync();
++			if (ret)
++				goto exit_err;
++		}
++
++		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
++			s3c24xx_snd_rxctrl(1);
++		else
++			s3c24xx_snd_txctrl(1);
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
++			s3c24xx_snd_rxctrl(0);
++		else
++			s3c24xx_snd_txctrl(0);
++		break;
++	default:
++		ret = -EINVAL;
++	}
++
++exit_err:
++	return ret;
++}
++
++static void s3c24xx_i2s_shutdown(struct snd_pcm_substream *substream)
++{
++	unsigned long iismod, iiscon, iisfcon;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++}
++
++#ifdef CONFIG_PM
++static int s3c24xx_i2s_suspend(struct platform_device *dev,
++	struct snd_soc_cpu_dai *dai)
++{
++}
++
++static int s3c24xx_i2s_resume(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++}
++
++#else
++#define s3c24xx_i2s_suspend	NULL
++#define s3c24xx_i2s_resume	NULL
++#endif
++
++/*
++ * Set S3C24xx Clock source
++ */
++static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
++	int clk_id, unsigned int freq, int dir)
++{
++	unsigned long iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	iismod &= ~S3C2440_IISMOD_MPLL;
++
++	switch (clk_id) {
++	case S3C24XX_CLKSRC_PCLK:
++		break;
++	case S3C24XX_CLKSRC_MPLL:
++		iismod |= S3C2440_IISMOD_MPLL;
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
++	return 0;
++}
++
++/*
++ * Set S3C24xx Clock dividers
++ */
++static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
++	int div_id, int div)
++{
++	unsigned long reg;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	switch (div_id) {
++	case S3C24XX_DIV_MCLK:
++		reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK;
++		writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
++		break;
++	case S3C24XX_DIV_BCLK:
++		reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS);
++		writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
++		break;
++	case S3C24XX_DIV_PRESCALER:
++		writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR);
++		reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
++		writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++
++/* To avoid duplicating clock code, allow machine driver to
++ * get the clockrate from here.
++ */
++
++unsigned long s3c24xx_i2s_get_clockrate(void)
++{
++	return clk_get_rate(s3c24xx_i2s.iis_clk);
++}
++
++EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
++
++static unsigned int s3c24xx_i2s_probe(struct platform_device *pdev,
++	struct snd_soc_cpu_dai *dai)
++{
++	DBG("Entered %s\n", __FUNCTION__);
++
++	s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
++	if (s3c24xx_i2s.regs == NULL)
++		return -ENXIO;
++
++	s3c24xx_i2s.iis_clk=clk_get(pdev, "iis");
++	if (s3c24xx_i2s.iis_clk == NULL) {
++		DBG("failed to get iis_clock\n");
++		return -ENODEV;
++	}
++	clk_enable(s3c24xx_i2s.iis_clk);
++
++	/* Configure the I2S pins in correct mode */
++	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
++	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
++	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
++	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
++	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
++
++	writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON);
++
++	s3c24xx_snd_txctrl(0);
++	s3c24xx_snd_rxctrl(0);
++
++	return 0;
++}
++
++#define S3C24XX_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)
++
++struct snd_soc_cpu_dai s3c24xx_i2s_dai = {
++	.name = "s3c24xx-i2s",
++	.id = 0,
++	.type = SND_SOC_DAI_I2S,
++	.probe = s3c24xx_i2s_probe,
++	.suspend = s3c24xx_i2s_suspend,
++	.resume = s3c24xx_i2s_resume,
++	.playback = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = S3C24XX_I2S_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = S3C24XX_I2S_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.startup = s3c24xx_i2s_startup,
++		.shutdown = s3c24xx_i2s_shutdown,
++		.trigger = s3c24xx_i2s_trigger,
++		.hw_params = s3c24xx_i2s_hw_params,},
++	.dai_ops = {
++		.set_fmt = s3c24xx_i2s_set_fmt,
++		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
++		.set_sysclk = s3c24xx_i2s_set_sysclk,
++	},
++};
++
++EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
++
++/* Module information */
++MODULE_AUTHOR("Ben Dooks, <ben at simtec.co.uk>");
++MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,35 @@
++/*
++ * s3c24xx-i2s.c  --  ALSA Soc Audio Layer
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Graeme Gregory
++ *         graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    10th Nov 2006   Initial version.
++ */
++
++#ifndef S3C24XXI2S_H_
++#define S3C24XXI2S_H_
++
++/* clock sources */
++#define S3C24XX_CLKSRC_PCLK 0
++#define S3C24XX_CLKSRC_MPLL 1
++
++/* Clock dividers */
++#define S3C24XX_DIV_MCLK	0
++#define S3C24XX_DIV_BCLK	1
++#define S3C24XX_DIV_PRESCALER	2
++
++/* prescaler */
++#define S3C24XX_PRESCALE(a,b) \
++	((a << S3C2410_IISPSR_INTSHIFT) | (b << S3C2410_IISPSR_EXTSHFIT))
++
++unsigned long s3c24xx_i2s_get_clockrate(void);
++
++#endif /*S3C24XXI2S_H_*/
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,484 @@
++/*
++ * s3c24xx-pcm.c  --  ALSA Soc Audio Layer
++ *
++ * (c) 2006 Wolfson Microelectronics PLC.
++ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ * (c) 2004-2005 Simtec Electronics
++ *	http://armlinux.simtec.co.uk/
++ *	Ben Dooks <ben at simtec.co.uk>
++ *
++ *  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
++ *    11th Dec 2006   Merged with Simtec driver
++ *    10th Nov 2006   Initial version.
++ */
++
++#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/io.h>
++#include <asm/hardware.h>
++#include <asm/arch/dma.h>
++#include <asm/arch/audio.h>
++
++#include "s3c24xx-pcm.h"
++
++#define S3C24XX_PCM_DEBUG 0
++#if S3C24XX_PCM_DEBUG
++#define DBG(x...) printk(KERN_DEBUG x)
++#else
++#define DBG(x...)
++#endif
++
++static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
++	.info				= ( SNDRV_PCM_INFO_INTERLEAVED |
++				    	SNDRV_PCM_INFO_BLOCK_TRANSFER |
++				    	SNDRV_PCM_INFO_MMAP |
++				    	SNDRV_PCM_INFO_MMAP_VALID |
++				    	0),
++	.formats			= ( SNDRV_PCM_FMTBIT_S16_LE |
++				    	SNDRV_PCM_FMTBIT_U16_LE |
++				    	SNDRV_PCM_FMTBIT_U8 |
++				    	SNDRV_PCM_FMTBIT_S8 ),
++	.rates				= SNDRV_PCM_RATE_44100,
++	.rate_min			= 8000,
++	.rate_max			= 48000,
++	.channels_min		= 2,
++	.channels_max		= 2,
++	.buffer_bytes_max	= 128*1024,
++	.period_bytes_min	= PAGE_SIZE,
++	.period_bytes_max	= PAGE_SIZE*2,
++	.periods_min		= 2,
++	.periods_max		= 128,
++	.fifo_size			= 32,
++};
++
++struct s3c24xx_runtime_data {
++	spinlock_t					lock;
++	int							state;
++	unsigned int		 		dma_loaded;
++	unsigned int		 		dma_limit;
++	unsigned int		 		dma_period;
++	dma_addr_t		 			dma_start;
++	dma_addr_t		 			dma_pos;
++	dma_addr_t		 			dma_end;
++	s3c24xx_pcm_dma_params_t 	*params;
++};
++
++/* s3c24xx_pcm_enqueue
++ *
++ * place a dma buffer onto the queue for the dma system
++ * to handle.
++*/
++
++static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
++{
++	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
++	dma_addr_t pos = prtd->dma_pos;
++	int ret;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	while ( prtd->dma_loaded < prtd->dma_limit) {
++		unsigned long len = prtd->dma_period;
++
++		DBG("dma_loaded: %d\n",prtd->dma_loaded);
++
++		if ((pos + len) > prtd->dma_end) {
++			len  = prtd->dma_end - pos;
++			DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
++			       __FUNCTION__, len);
++		}
++
++		ret = s3c2410_dma_enqueue(prtd->params->channel, substream, pos, len);
++
++		if (ret == 0) {
++			prtd->dma_loaded++;
++			pos += prtd->dma_period;
++			if (pos >= prtd->dma_end)
++				pos = prtd->dma_start;
++		} else
++			break;
++	}
++
++	prtd->dma_pos = pos;
++}
++
++void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
++							void *dev_id, int size,
++							enum s3c2410_dma_buffresult result)
++{
++	struct snd_pcm_substream *substream = dev_id;
++	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
++		return;
++
++	if (substream)
++		snd_pcm_period_elapsed(substream);
++
++	spin_lock(&prtd->lock);
++	if (prtd->state & ST_RUNNING) {
++		prtd->dma_loaded--;
++		s3c24xx_pcm_enqueue(substream);
++	}
++
++	spin_unlock(&prtd->lock);
++}
++
++static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct s3c24xx_runtime_data *prtd = runtime->private_data;
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	s3c24xx_pcm_dma_params_t *dma = rtd->dai->cpu_dai->dma_data;
++	unsigned long totbytes = params_buffer_bytes(params);
++	unsigned long flags;
++	int ret=0;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	/* prepare DMA */
++	prtd->params = dma;
++
++	DBG("params %p, client %p, channel %d\n", prtd->params,
++		prtd->params->client, prtd->params->channel);
++
++	ret = s3c2410_dma_request(prtd->params->channel,
++				  prtd->params->client, NULL);
++
++	if (ret) {
++		DBG(KERN_ERR "failed to get dma channel\n");
++		return ret;
++	}
++
++	/* channel needs configuring for mem=>device, increment memory addr,
++	 * sync to pclk, half-word transfers to the IIS-FIFO. */
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		s3c2410_dma_devconfig(prtd->params->channel,
++						S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
++						S3C2410_DISRCC_APB, prtd->params->dma_addr);
++
++		s3c2410_dma_config(prtd->params->channel,
++						2, S3C2410_DCON_SYNC_PCLK | S3C2410_DCON_HANDSHAKE);
++	} else {
++		s3c2410_dma_config(prtd->params->channel,
++						2, S3C2410_DCON_HANDSHAKE | S3C2410_DCON_SYNC_PCLK);
++
++		s3c2410_dma_devconfig(prtd->params->channel,
++						S3C2410_DMASRC_HW, 0x3,
++						prtd->params->dma_addr);
++	}
++
++	s3c2410_dma_set_buffdone_fn(prtd->params->channel,
++				    s3c24xx_audio_buffdone);
++
++	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
++
++	runtime->dma_bytes = totbytes;
++
++	spin_lock_irqsave(&prtd->lock, flags);
++	prtd->dma_loaded	= 0;
++	prtd->dma_limit	= runtime->hw.periods_min;
++	prtd->dma_period	= params_period_bytes(params);
++	prtd->dma_start	= runtime->dma_addr;
++	prtd->dma_pos	= prtd->dma_start;
++	prtd->dma_end	= prtd->dma_start + totbytes;
++	spin_unlock_irqrestore(&prtd->lock, flags);
++
++	return 0;
++}
++
++static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	// TODO - do we need to ensure DMA flushed
++	snd_pcm_set_runtime_buffer(substream, NULL);
++
++	if(prtd->params) {
++		s3c2410_dma_free(prtd->params->channel, prtd->params->client);
++		prtd->params = NULL;
++	}
++
++}
++
++static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
++{
++	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	int ret = 0;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	/* flush the DMA channel */
++
++	s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH);
++	prtd->dma_loaded = 0;
++	prtd->dma_pos = prtd->dma_start;
++
++	/* enqueue dma buffers */
++	s3c24xx_pcm_enqueue(substream);
++
++	return ret;
++}
++
++static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
++	int ret = 0;
++	unsigned long flags;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	spin_lock_irqsave(&prtd->lock, flags);
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++	case SNDRV_PCM_TRIGGER_RESUME:
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		prtd->state |= ST_RUNNING;
++		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
++		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
++		break;
++
++	case SNDRV_PCM_TRIGGER_STOP:
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		prtd->state &= ~ST_RUNNING;
++		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP);
++		break;
++
++	default:
++		ret = -EINVAL;
++	}
++
++	spin_unlock_irqrestore(&prtd->lock, flags);
++
++	return ret;
++}
++
++static snd_pcm_uframes_t s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct s3c24xx_runtime_data *prtd = runtime->private_data;
++	unsigned long flags;
++	unsigned long res;
++	dma_addr_t src, dst;
++	snd_pcm_uframes_t ret;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	spin_lock_irqsave(&prtd->lock, flags);
++	s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
++
++	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
++		res = dst - prtd->dma_start;
++	else
++		res = src - prtd->dma_start;
++
++	spin_unlock_irqrestore(&prtd->lock, flags);
++
++	DBG("Pointer %x %x\n",src,dst);
++
++	/* we seem to be getting the odd error from the pcm library due
++	 * to out-of-bounds pointers. this is maybe due to the dma engine
++	 * not having loaded the new values for the channel before being
++	 * callled... (todo - fix )
++	 */
++
++	if (res > (prtd->dma_end - prtd->dma_start))
++	//	DBG("%s: %lx,%lx (%lx) from %lx,%lx\n", __FUNCTION__,
++	//	    (long)src, (long)dst, res,
++	//	    (long)prtd->dma_start, (long)prtd->dma_end);
++
++	if (res >= snd_pcm_lib_buffer_bytes(substream)) {
++	//	DBG("%s: %lx,%lx (%lx) from %lx,%lx\n", __FUNCTION__,
++	//	    (long)src, (long)dst, res,
++	//	    (long)prtd->dma_start, (long)prtd->dma_end);
++
++	//	if (res > snd_pcm_lib_buffer_bytes(substream))
++	//		DBG("%s: res %lx >= %lx\n", __FUNCTION__,
++	//		    res, (long)snd_pcm_lib_buffer_bytes(substream));
++
++		if (res == snd_pcm_lib_buffer_bytes(substream))
++			res = 0;
++	}
++
++	ret = bytes_to_frames(substream->runtime, res);
++
++	return ret;
++
++}
++
++static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct s3c24xx_runtime_data *prtd;
++
++	int ret;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
++
++	if((prtd = kzalloc(sizeof(struct s3c24xx_runtime_data),
++		GFP_KERNEL)) == NULL) {
++		ret = -ENOMEM;
++		goto out;
++	}
++
++	runtime->private_data = prtd;
++	return 0;
++
++out:
++	return ret;
++}
++
++static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct s3c24xx_runtime_data *prtd = runtime->private_data;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	if(prtd)
++		kfree(prtd);
++	else {
++		DBG("s3c24xx_pcm_close called with prtd == NULL\n");
++	}
++
++	return 0;
++}
++
++static int
++s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
++                                     runtime->dma_area,
++                                     runtime->dma_addr,
++                                     runtime->dma_bytes);
++}
++
++struct snd_pcm_ops s3c24xx_pcm_ops = {
++	.open		= s3c24xx_pcm_open,
++	.close		= s3c24xx_pcm_close,
++	.ioctl		= snd_pcm_lib_ioctl,
++	.hw_params	= s3c24xx_pcm_hw_params,
++	.hw_free	= s3c24xx_pcm_hw_free,
++	.prepare	= s3c24xx_pcm_prepare,
++	.trigger	= s3c24xx_pcm_trigger,
++	.pointer	= s3c24xx_pcm_pointer,
++	.mmap		= s3c24xx_pcm_mmap,
++};
++
++static int s3c24xx_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 = s3c24xx_pcm_hardware.buffer_bytes_max;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	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 s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
++{
++	struct snd_pcm_substream *substream;
++	struct snd_dma_buffer *buf;
++	int stream;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	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 s3c24xx_pcm_dmamask = 0xffffffff;
++
++int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
++	struct snd_pcm *pcm)
++{
++	int ret = 0;
++
++	DBG("Entered %s\n", __FUNCTION__);
++
++	if (!card->dev->dma_mask)
++		card->dev->dma_mask = &s3c24xx_pcm_dmamask;
++	if (!card->dev->coherent_dma_mask)
++		card->dev->coherent_dma_mask = 0xffffffff;
++
++	if (dai->playback.channels_min) {
++		ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
++			SNDRV_PCM_STREAM_PLAYBACK);
++		if (ret)
++			goto out;
++	}
++
++	if (dai->capture.channels_min) {
++		ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
++			SNDRV_PCM_STREAM_CAPTURE);
++		if (ret)
++			goto out;
++	}
++ out:
++	return ret;
++}
++
++struct snd_soc_platform s3c24xx_soc_platform = {
++	.name		= "s3c24xx-audio",
++	.pcm_ops 	= &s3c24xx_pcm_ops,
++	.pcm_new	= s3c24xx_pcm_new,
++	.pcm_free	= s3c24xx_pcm_free_dma_buffers,
++};
++
++EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
++
++MODULE_AUTHOR("Ben Dooks, <ben at simtec.co.uk>");
++MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.h
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.h	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,33 @@
++/*
++ * 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
++
++#define ST_RUNNING		(1<<0)
++#define ST_OPENED		(1<<1)
++
++typedef struct {
++	struct s3c2410_dma_client *client;			/* stream identifier */
++	int channel;						/* Channel ID */
++	dma_addr_t dma_addr;
++} s3c24xx_pcm_dma_params_t;
++
++#define S3C24XX_DAI_I2S			0
++
++extern struct snd_soc_cpu_dai s3c24xx_i2s_dai;
++
++/* platform data */
++extern struct snd_soc_platform s3c24xx_soc_platform;
++extern struct snd_ac97_bus_ops s3c24xx_ac97_ops;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440.c	2007-01-24 12:19:24.000000000 +0100
+@@ -0,0 +1,320 @@
++/*
++ * smdk2440.c  --  ALSA Soc Audio Layer
++ *
++ * (c) 2006 Wolfson Microelectronics PLC.
++ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ * (c) 2004-2005 Simtec Electronics
++ *	http://armlinux.simtec.co.uk/
++ *	Ben Dooks <ben at simtec.co.uk>
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ * This module is a modified version of the s3c24xx I2S driver supplied by
++ * Ben Dooks of Simtec and rejigged to the ASoC style at Wolfson Microelectronics
++ *
++ *  Revision history
++ *    11th Dec 2006   Merged with Simtec driver
++ *    10th Nov 2006   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/regs-iis.h>
++#include <asm/arch/regs-clock.h>
++#include <asm/arch/regs-gpio.h>
++#include <asm/arch/hardware.h>
++#include <asm/arch/audio.h>
++#include <asm/io.h>
++#include <asm/arch/spi-gpio.h>
++#include "../codecs/uda1380.h"
++#include "s3c24xx-pcm.h"
++#include "s3c24xx-i2s.h"
++
++#define SMDK2440_DEBUG 0
++#if SMDK2440_DEBUG
++#define DBG(x...) printk(KERN_DEBUG x)
++#else
++#define DBG(x...)
++#endif
++
++/* audio clock in Hz */
++#define SMDK_CLOCK_SOURCE S3C24XX_CLKSRC_MPLL
++#define SMDK_CRYSTAL_CLOCK 16934400
++
++static int smdk2440_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec *codec = rtd->socdev->codec;
++
++	DBG("Entered smdk2440_startup\n");
++
++	return 0;
++}
++
++static int smdk2440_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec *codec = rtd->socdev->codec;
++
++	DBG("Entered smdk2440_shutdown\n");
++
++	return 0;
++}
++
++static int smdk2440_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	unsigned long iis_clkrate;
++	int div, div256, div384, diff256, diff384, bclk, mclk;
++	int ret;
++	unsigned int rate=params_rate(params);
++
++	DBG("Entered %s\n",__FUNCTION__);
++
++	iis_clkrate = s3c24xx_i2s_get_clockrate();
++
++	/* Using PCLK doesnt seem to suit audio particularly well on these cpu's
++	 */
++
++	div256 = iis_clkrate / (rate * 256);
++	div384 = iis_clkrate / (rate * 384);
++
++	if (((iis_clkrate / div256) - (rate * 256)) <
++		((rate * 256) - (iis_clkrate / (div256 + 1)))) {
++		diff256 = (iis_clkrate / div256) - (rate * 256);
++	} else {
++		div256++;
++		diff256 = (iis_clkrate / div256) - (rate * 256);
++	}
++
++	if (((iis_clkrate / div384) - (rate * 384)) &