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diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
new file mode 100644
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--- /dev/null
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+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/p/en_US/business/design
+
+
+I2S
+===
+
+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmission, 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 LRCLKs, this allows for simultaneous 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 BCLKs before LRC
+                     transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, which 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 simultaneously (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.
diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
new file mode 100644
index 00000000..b1300162
--- /dev/null
+++ b/Documentation/sound/alsa/soc/clocking.txt
@@ -0,0 +1,51 @@
+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 referred 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. PLLs and CPU based clocks) are configurable in that
+their speed can be altered by software (depending on the system use and to save
+power). Other master clocks are fixed at a 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 is best to configure BCLK to the lowest possible speed (depending on your
+rate, number of channels and word size) to save on power.
+
+It is also desirable to use the codec (if possible) to drive (or master) the
+audio clocks as it usually gives more accurate sample rates than the CPU.
+
+
+
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
new file mode 100644
index 00000000..bce23a4a
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,195 @@
+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) Codec DAI and PCM configuration
+ 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
+ 3) Mixers and audio controls
+ 4) Codec audio operations
+
+Optionally, codec drivers can also provide:-
+
+ 5) DAPM description.
+ 6) DAPM event handler.
+ 7) DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+1 - Codec DAI and PCM configuration
+-----------------------------------
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+	.name = "wm8731-hifi",
+	.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 = &wm8731_dai_ops,
+	.symmetric_rates = 1,
+};
+
+
+2 - Codec control IO
+--------------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec drivers 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;
+
+
+3 - 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
+
+
+4 - Codec Audio Operations
+--------------------------
+The codec driver also supports the following ALSA operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+};
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+
+5 - DAPM description.
+---------------------
+The Dynamic Audio Power Management description describes the codec power
+components and 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.
+
+
+6 - 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 is 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 */
+
+
+7 - Codec DAC digital mute control
+----------------------------------
+Most codecs have a digital mute before the DACs 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_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+	if (mute)
+		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+	else
+		snd_soc_write(codec, WM8974_DAC, mute_reg);
+	return 0;
+}
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
new file mode 100644
index 00000000..05bf5a0e
--- /dev/null
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -0,0 +1,294 @@
+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 decisions 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 - DACs and ADCs.
+      Enabled and disabled when stream playback/capture is started and
+      stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching decisions 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 one of many 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 Supply     - Power or clock supply widget used by other widgets.
+ 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
+convenience 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 ADCs
+(analog to digital converters) and DACs (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 codec
+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 affect 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)),
+
+If you dont want the mixer elements prefixed with the name of the mixer widget,
+you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
+as for SND_SOC_DAPM_MIXER.
+
+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)
+{
+	gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+	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 soft power control. In this case it is 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.
+
+e.g., from the WM8731 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 its kcontrol name. We can now
+connect the destination widget (wrt audio signal) with its 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)
+{
+	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+	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 */
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
new file mode 100644
index 00000000..fcf82a41
--- /dev/null
+++ b/Documentation/sound/alsa/soc/jack.txt
@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h.  ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+   user visible jack.  In embedded systems it is common for multiple
+   to be present on a single jack but handled by separate bits of
+   hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+   automatically based on the detected jack status (eg, turning off the
+   headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone.  Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space.  The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack.  Each snd_soc_jack has zero or more of these
+which are updated automatically.  They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins().  The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update.  The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function.  Other methods are
+also available, for example integrated into CODECs.  One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware.  The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
new file mode 100644
index 00000000..d50c14df
--- /dev/null
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -0,0 +1,91 @@
+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_card {
+	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 DAIs 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;
+
+	...
+};
+
+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, DAIs
+and DMA is suspended and resumed. Optional.
+
+
+Machine DAI Configuration
+-------------------------
+The machine DAI configuration glues all the codec and CPU DAIs 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, unconnected codec pins can be set as such.
+
+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_name = "pxa-is2-dai",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8713-codec.0-001a",
+	.init = corgi_wm8731_init,
+	.ops = &corgi_ops,
+};
+
+struct snd_soc_card then sets up the machine with its DAIs. e.g.
+
+/* corgi audio machine driver */
+static struct snd_soc_card snd_soc_corgi = {
+	.name = "Corgi",
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+};
+
+
+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.
+
+
+Machine Controls
+----------------
+
+Machine specific audio mixer controls can be added in the DAI init function.
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
new file mode 100644
index 00000000..138ac88c
--- /dev/null
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -0,0 +1,86 @@
+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 processors (e.g.
+pxa2xx, au1x00, iMX, etc) and portable audio codecs.  Prior to the ASoC
+subsystem there was some support in the kernel for SoC audio, however it
+had some limitations:-
+
+  * Codec drivers were often tightly coupled to the underlying SoC
+    CPU. This is not ideal and leads to code duplication - for example,
+    Linux had different wm8731 drivers for 4 different SoC platforms.
+
+  * There was 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 often require
+    machine specific code to re-route audio, enable amps, etc., after such an
+    event.
+
+  * Drivers tended 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 was 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 its audio interface capabilities with the
+    core and are subsequently matched and configured when the application
+    hardware parameters are known.
+
+  * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+    its 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 amplifier).
+
+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 (e.g. turning 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.
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
new file mode 100644
index 00000000..d57efad3
--- /dev/null
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -0,0 +1,66 @@
+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)(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);
+};
+
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
+
+struct snd_soc_platform_driver {
+	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 *);
+
+	/*
+	 * For platform caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+
+	/* platform stream ops */
+	struct snd_pcm_ops *pcm_ops;
+};
+
+Please refer to the ALSA driver documentation for details of audio DMA.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+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.
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
new file mode 100644
index 00000000..e1e74daa
--- /dev/null
+++ b/Documentation/sound/alsa/soc/pops_clicks.txt
@@ -0,0 +1,52 @@
+Audio Pops and Clicks
+=====================
+
+Pops and clicks are unwanted audio artifacts caused by the powering up and down
+of components within the audio subsystem. This is noticeable on PCs 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
+currently, however future audio codec hardware will have better pop and click
+suppression.  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 occurred. 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.