Adding upstream version 5.10.209.upstream/5.10.209

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

12 files changed, 1552 insertions, 0 deletions

diff --git a/Documentation/sound/soc/clocking.rst b/Documentation/sound/soc/clocking.rst
new file mode 100644
index 000000000..32122d687
--- /dev/null
+++ b/Documentation/sound/soc/clocking.rst
@@ -0,0 +1,46 @@
+==============
+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/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
new file mode 100644
index 000000000..4eaa9a0c4
--- /dev/null
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -0,0 +1,113 @@
+==============================================
+Creating codec to codec dai link for ALSA dapm
+==============================================
+
+Mostly the flow of audio is always from CPU to codec so your system
+will look as below:
+::
+
+   ---------          ---------
+  |         |  dai   |         |
+      CPU    ------->    codec
+  |         |        |         |
+   ---------          ---------
+
+In case your system looks as below:
+::
+
+                       ---------
+                      |         |
+                        codec-2
+                      |         |
+                      ---------
+                           |
+                         dai-2
+                           |
+   ----------          ---------
+  |          |  dai-1 |         |
+      CPU     ------->  codec-1
+  |          |        |         |
+   ----------          ---------
+                           |
+                         dai-3
+                           |
+                       ---------
+                      |         |
+                        codec-3
+                      |         |
+                       ---------
+
+Suppose codec-2 is a bluetooth chip and codec-3 is connected to
+a speaker and you have a below scenario:
+codec-2 will receive the audio data and the user wants to play that
+audio through codec-3 without involving the CPU.This
+aforementioned case is the ideal case when codec to codec
+connection should be used.
+
+Your dai_link should appear as below in your machine
+file:
+::
+
+ /*
+  * this pcm stream only supports 24 bit, 2 channel and
+  * 48k sampling rate.
+  */
+ static const struct snd_soc_pcm_stream dsp_codec_params = {
+        .formats = SNDRV_PCM_FMTBIT_S24_LE,
+        .rate_min = 48000,
+        .rate_max = 48000,
+        .channels_min = 2,
+        .channels_max = 2,
+ };
+
+ {
+    .name = "CPU-DSP",
+    .stream_name = "CPU-DSP",
+    .cpu_dai_name = "samsung-i2s.0",
+    .codec_name = "codec-2,
+    .codec_dai_name = "codec-2-dai_name",
+    .platform_name = "samsung-i2s.0",
+    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+            | SND_SOC_DAIFMT_CBM_CFM,
+    .ignore_suspend = 1,
+    .params = &dsp_codec_params,
+ },
+ {
+    .name = "DSP-CODEC",
+    .stream_name = "DSP-CODEC",
+    .cpu_dai_name = "wm0010-sdi2",
+    .codec_name = "codec-3,
+    .codec_dai_name = "codec-3-dai_name",
+    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+            | SND_SOC_DAIFMT_CBM_CFM,
+    .ignore_suspend = 1,
+    .params = &dsp_codec_params,
+ },
+
+Above code snippet is motivated from sound/soc/samsung/speyside.c.
+
+Note the "params" callback which lets the dapm know that this
+dai_link is a codec to codec connection.
+
+In dapm core a route is created between cpu_dai playback widget
+and codec_dai capture widget for playback path and vice-versa is
+true for capture path. In order for this aforementioned route to get
+triggered, DAPM needs to find a valid endpoint which could be either
+a sink or source widget corresponding to playback and capture path
+respectively.
+
+In order to trigger this dai_link widget, a thin codec driver for
+the speaker amp can be created as demonstrated in wm8727.c file, it
+sets appropriate constraints for the device even if it needs no control.
+
+Make sure to name your corresponding cpu and codec playback and capture
+dai names ending with "Playback" and "Capture" respectively as dapm core
+will link and power those dais based on the name.
+
+A dai_link in a "simple-audio-card" will automatically be detected as
+codec to codec when all DAIs on the link belong to codec components.
+The dai_link will be initialized with the subset of stream parameters
+(channels, format, sample rate) supported by all DAIs on the link. Since
+there is no way to provide these parameters in the device tree, this is
+mostly useful for communication with simple fixed-function codecs, such
+as a Bluetooth controller or cellular modem.
diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst
new file mode 100644
index 000000000..8a9737eb7
--- /dev/null
+++ b/Documentation/sound/soc/codec.rst
@@ -0,0 +1,190 @@
+=======================
+ASoC Codec Class Driver
+=======================
+
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP 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 class driver *must* provide the following features:-
+
+1. Codec DAI and PCM configuration
+2. Codec control IO - using RegMap API
+3. Mixers and audio controls
+4. Codec audio operations
+5. DAPM description.
+6. DAPM event handler.
+
+Optionally, codec drivers can also provide:-
+
+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
+===========================
+
+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,
+  };
+
+
+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 driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
+
+
+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
+
+
+Codec Audio Operations
+----------------------
+The codec driver also supports the following ALSA PCM 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/
+
+
+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.rst for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+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 */
+
+
+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_component *component = dai->component;
+	u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
+
+	if (mute)
+		snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
+	else
+		snd_soc_component_write(component, WM8974_DAC, mute_reg);
+	return 0;
+  }
diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
new file mode 100644
index 000000000..009b07e5a
--- /dev/null
+++ b/Documentation/sound/soc/dai.rst
@@ -0,0 +1,64 @@
+==================================
+ASoC Digital Audio Interface (DAI)
+==================================
+
+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 :
+https://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, while 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:-
+
+I2S
+  MSB is transmitted on the falling edge of the first BCLK after LRC
+  transition.
+
+Left Justified
+  MSB is transmitted on transition of LRC.
+
+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 while the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+while 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:-
+
+Mode A
+  MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+Mode B
+  MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
new file mode 100644
index 000000000..8e4410793
--- /dev/null
+++ b/Documentation/sound/soc/dapm.rst
@@ -0,0 +1,360 @@
+===================================================
+Dynamic Audio Power Management for Portable Devices
+===================================================
+
+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
+
+Codec bias 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.
+
+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
+
+Path domain
+      audio subsystem signal paths
+
+      Automatically set when mixer and mux settings are changed by the user.
+      e.g. alsamixer, amixer.
+
+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.
+
+
+DAPM Widgets
+============
+
+Audio DAPM widgets fall into a number of types:-
+
+Mixer
+	Mixes several analog signals into a single analog signal.
+Mux
+	An analog switch that outputs only one of many inputs.
+PGA
+	A programmable gain amplifier or attenuation widget.
+ADC
+	Analog to Digital Converter
+DAC
+	Digital to Analog Converter
+Switch
+	An analog switch
+Input
+	A codec input pin
+Output
+	A codec output pin
+Headphone
+	Headphone (and optional Jack)
+Mic
+	Mic (and optional Jack)
+Line
+	Line Input/Output (and optional Jack)
+Speaker
+	Speaker
+Supply
+	Power or clock supply widget used by other widgets.
+Regulator
+	External regulator that supplies power to audio components.
+Clock
+	External clock that supplies clock to audio components.
+AIF IN
+	Audio Interface Input (with TDM slot mask).
+AIF OUT
+	Audio Interface Output (with TDM slot mask).
+Siggen
+	Signal Generator.
+DAI IN
+	Digital Audio Interface Input.
+DAI OUT
+	Digital Audio Interface Output.
+DAI Link
+	DAI Link between two DAI structures
+Pre
+	Special PRE widget (exec before all others)
+Post
+	Special POST widget (exec after all others)
+Buffer
+	Inter widget audio data buffer within a DSP.
+Scheduler
+	DSP internal scheduler that schedules component/pipeline processing
+	work.
+Effect
+	Widget that performs an audio processing effect.
+SRC
+	Sample Rate Converter within DSP or CODEC
+ASRC
+	Asynchronous Sample Rate Converter within DSP or CODEC
+Encoder
+	Widget that encodes audio data from one format (usually PCM) to another
+	usually more compressed format.
+Decoder
+	Widget that decodes audio data from a compressed format to an
+	uncompressed format like PCM.
+
+
+(Widgets are defined in include/sound/soc-dapm.h)
+
+Widgets can be added to the sound card by any of the component driver types.
+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.
+
+
+Stream Domain Widgets
+---------------------
+
+Stream Widgets relate to the stream power domain and only consist of ADCs
+(analog to digital converters), DACs (digital to analog converters),
+AIF IN and AIF OUT.
+
+Stream widgets have the following format:-
+::
+
+  SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+  SND_SOC_DAPM_AIF_IN(name, stream, slot, 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),
+
+e.g. stream widgets for AIF
+::
+
+  SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+  SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+
+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 don't 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.
+
+
+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 or DSP) that can be independently
+powered. e.g.
+
+* Speaker Amp
+* Microphone Bias
+* 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),
+
+
+Codec (BIAS) Domain
+-------------------
+
+The codec bias 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.
+
+
+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().
+
+
+Codec/DSP Widget Interconnections
+=================================
+
+Widgets are connected to each other within the codec, platform 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 or DSP (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.
+
+
+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.
+
+
+Endpoint Widgets
+================
+An endpoint is a start or end point (widget) of an audio signal within the
+machine and includes the codec. e.g.
+
+* Headphone Jack
+* Internal Speaker
+* Internal Mic
+* Mic Jack
+* Codec Pins
+
+Endpoints are added to the DAPM graph so that their usage can be determined in
+order to save power. e.g. NC codecs pins will be switched OFF, unconnected
+jacks can also be switched OFF.
+
+
+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.
+
+
+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/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst
new file mode 100644
index 000000000..77f67ded5
--- /dev/null
+++ b/Documentation/sound/soc/dpcm.rst
@@ -0,0 +1,388 @@
+===========
+Dynamic PCM
+===========
+
+Description
+===========
+
+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
+drivers that expose several ALSA PCMs and can route to multiple DAIs.
+
+The DPCM runtime routing is determined by the ALSA mixer settings in the same
+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
+graph representing the DSP internal audio paths and uses the mixer settings to
+determine the path used by each ALSA PCM.
+
+DPCM re-uses all the existing component codec, platform and DAI drivers without
+any modifications.
+
+
+Phone Audio System with SoC based DSP
+-------------------------------------
+
+Consider the following phone audio subsystem. This will be used in this
+document for all examples :-
+::
+
+  | Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+  
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
+
+
+
+Example - DPCM Switching playback from DAI0 to DAI1
+---------------------------------------------------
+
+Audio is being played to the Headset. After a while the user removes the headset
+and audio continues playing on the speakers.
+
+Playback on PCM0 to Headset would look like :-
+::
+
+                      *************
+  PCM0 <============> *           * <====DAI0=====> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+The headset is removed from the jack by user so the speakers must now be used :-
+::
+
+                      *************
+  PCM0 <============> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <====DAI1=====> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+The audio driver processes this as follows :-
+
+1. Machine driver receives Jack removal event.
+
+2. Machine driver OR audio HAL disables the Headset path.
+
+3. DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
+   for headset since the path is now disabled.
+
+4. Machine driver or audio HAL enables the speaker path.
+
+5. DPCM runs the PCM ops for startup(), hw_params(), prepare() and
+   trigger(start) for DAI1 Speakers since the path is enabled.
+
+In this example, the machine driver or userspace audio HAL can alter the routing
+and then DPCM will take care of managing the DAI PCM operations to either bring
+the link up or down. Audio playback does not stop during this transition.
+
+
+
+DPCM machine driver
+===================
+
+The DPCM enabled ASoC machine driver is similar to normal machine drivers
+except that we also have to :-
+
+1. Define the FE and BE DAI links.
+
+2. Define any FE/BE PCM operations.
+
+3. Define widget graph connections.
+
+
+FE and BE DAI links
+-------------------
+::
+
+  | Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+  
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
+FE DAI links are defined as follows :-
+::
+
+  static struct snd_soc_dai_link machine_dais[] = {
+	{
+		.name = "PCM0 System",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "dsp-audio",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dynamic = 1,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	.....< other FE and BE DAI links here >
+  };
+
+This FE DAI link is pretty similar to a regular DAI link except that we also
+set the DAI link to a DPCM FE with the ``dynamic = 1``. The supported FE stream
+directions should also be set with the ``dpcm_playback`` and ``dpcm_capture``
+flags. There is also an option to specify the ordering of the trigger call for
+each FE. This allows the ASoC core to trigger the DSP before or after the other
+components (as some DSPs have strong requirements for the ordering DAI/DSP
+start and stop sequences).
+
+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
+dynamic and will change depending on runtime config.
+
+The BE DAIs are configured as follows :-
+::
+
+  static struct snd_soc_dai_link machine_dais[] = {
+	.....< FE DAI links here >
+	{
+		.name = "Codec Headset",
+		.cpu_dai_name = "ssp-dai.0",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "rt5640.0-001c",
+		.codec_dai_name = "rt5640-aif1",
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = hswult_ssp0_fixup,
+		.ops = &haswell_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	.....< other BE DAI links here >
+  };
+
+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
+the ``no_pcm`` flag to mark it has a BE and sets flags for supported stream
+directions using ``dpcm_playback`` and ``dpcm_capture`` above.
+
+The BE has also flags set for ignoring suspend and PM down time. This allows
+the BE to work in a hostless mode where the host CPU is not transferring data
+like a BT phone call :-
+::
+
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <====DAI2=====> MODEM
+                      *           *
+  PCM3 <------------> *           * <====DAI3=====> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+This allows the host CPU to sleep while the DSP, MODEM DAI and the BT DAI are
+still in operation.
+
+A BE DAI link can also set the codec to a dummy device if the codec is a device
+that is managed externally.
+
+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
+DSP firmware.
+
+
+FE/BE PCM operations
+--------------------
+
+The BE above also exports some PCM operations and a ``fixup`` callback. The fixup
+callback is used by the machine driver to (re)configure the DAI based upon the
+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
+
+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
+DAI0. This means all FE hw_params have to be fixed in the machine driver for
+DAI0 so that the DAI is running at desired configuration regardless of the FE
+configuration.
+::
+
+  static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+  {
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set DAI0 to 16 bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+  }
+
+The other PCM operation are the same as for regular DAI links. Use as necessary.
+
+
+Widget graph connections
+------------------------
+
+The BE DAI links will normally be connected to the graph at initialisation time
+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
+has to be set explicitly in the driver :-
+::
+
+  /* BE for codec Headset -  DAI0 is dummy and managed by DSP FW */
+  {"DAI0 CODEC IN", NULL, "AIF1 Capture"},
+  {"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
+
+
+Writing a DPCM DSP driver
+=========================
+
+The DPCM DSP driver looks much like a standard platform class ASoC driver
+combined with elements from a codec class driver. A DSP platform driver must
+implement :-
+
+1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
+
+2. DAPM graph showing DSP audio routing from FE DAIs to BEs.
+
+3. DAPM widgets from DSP graph.
+
+4. Mixers for gains, routing, etc.
+
+5. DMA configuration.
+
+6. BE AIF widgets.
+
+Items 6 is important for routing the audio outside of the DSP. AIF need to be
+defined for each BE and each stream direction. e.g for BE DAI0 above we would
+have :-
+::
+
+  SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+  SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+The BE AIF are used to connect the DSP graph to the graphs for the other
+component drivers (e.g. codec graph).
+
+
+Hostless PCM streams
+====================
+
+A hostless PCM stream is a stream that is not routed through the host CPU. An
+example of this would be a phone call from handset to modem.
+::
+
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <====DAI1=====> Codec Speakers/Mic
+                      *   DSP     *
+  PCM2 <------------> *           * <====DAI2=====> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+In this case the PCM data is routed via the DSP. The host CPU in this use case
+is only used for control and can sleep during the runtime of the stream.
+
+The host can control the hostless link either by :-
+
+ 1. Configuring the link as a CODEC <-> CODEC style link. In this case the link
+    is enabled or disabled by the state of the DAPM graph. This usually means
+    there is a mixer control that can be used to connect or disconnect the path
+    between both DAIs.
+
+ 2. Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
+    graph. Control is then carried out by the FE as regular PCM operations.
+    This method gives more control over the DAI links, but requires much more
+    userspace code to control the link. Its recommended to use CODEC<->CODEC
+    unless your HW needs more fine grained sequencing of the PCM ops.
+
+
+CODEC <-> CODEC link
+--------------------
+
+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
+The machine driver sets some additional parameters to the DAI link i.e.
+::
+
+  static const struct snd_soc_pcm_stream dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+  };
+
+  static struct snd_soc_dai_link dais[] = {
+	< ... more DAI links above ... >
+	{
+		.name = "MODEM",
+		.stream_name = "MODEM",
+		.cpu_dai_name = "dai2",
+		.codec_dai_name = "modem-aif1",
+		.codec_name = "modem",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dai_params,
+	}
+	< ... more DAI links here ... >
+
+These parameters are used to configure the DAI hw_params() when DAPM detects a
+valid path and then calls the PCM operations to start the link. DAPM will also
+call the appropriate PCM operations to disable the DAI when the path is no
+longer valid.
+
+
+Hostless FE
+-----------
+
+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
+This means creating a new FE that is connected with a virtual path to both
+DAI links. The DAI links will be started when the FE PCM is started and stopped
+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
+this configuration.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
new file mode 100644
index 000000000..e57df2dab
--- /dev/null
+++ b/Documentation/sound/soc/index.rst
@@ -0,0 +1,20 @@
+==============
+ALSA SoC Layer
+==============
+
+The documentation is spilt into the following sections:-
+
+.. toctree::
+   :maxdepth: 2
+
+   overview
+   codec
+   dai
+   dapm
+   platform
+   machine
+   pops-clicks
+   clocking
+   jack
+   dpcm
+   codec-to-codec
diff --git a/Documentation/sound/soc/jack.rst b/Documentation/sound/soc/jack.rst
new file mode 100644
index 000000000..644b99ecb
--- /dev/null
+++ b/Documentation/sound/soc/jack.rst
@@ -0,0 +1,72 @@
+===================
+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/soc/machine.rst b/Documentation/sound/soc/machine.rst
new file mode 100644
index 000000000..515c9444d
--- /dev/null
+++ b/Documentation/sound/soc/machine.rst
@@ -0,0 +1,97 @@
+===================
+ASoC Machine Driver
+===================
+
+The ASoC machine (or board) driver is the code that glues together all the
+component drivers (e.g. codecs, platforms and DAIs). It also describes the
+relationships between each component which include audio paths, GPIOs,
+interrupts, clocking, jacks and voltage regulators.
+
+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/soc/overview.rst b/Documentation/sound/soc/overview.rst
new file mode 100644
index 000000000..dc8370bbf
--- /dev/null
+++ b/Documentation/sound/soc/overview.rst
@@ -0,0 +1,69 @@
+=======================
+ALSA SoC Layer Overview
+=======================
+
+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
+multiple re-usable component drivers :-
+
+  * Codec class drivers: The codec class driver is platform independent and
+    contains audio controls, audio interface capabilities, codec DAPM
+    definition and codec IO functions. This class extends to BT, FM and MODEM
+    ICs if required. Codec class drivers should be generic code that can run
+    on any architecture and machine.
+
+  * Platform class drivers: The platform class driver includes the audio DMA
+    engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
+    and any audio DSP drivers for that platform.
+
+  * Machine class driver: The machine driver class acts as the glue that
+    describes and binds the other component drivers together to form an ALSA
+    "sound card device". It handles any machine specific controls and
+    machine level audio events (e.g. turning on an amp at start of playback).
diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst
new file mode 100644
index 000000000..c1badea53
--- /dev/null
+++ b/Documentation/sound/soc/platform.rst
@@ -0,0 +1,78 @@
+====================
+ASoC Platform Driver
+====================
+
+An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
+drivers and DSP drivers. 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_component_driver:-
+::
+
+  struct snd_soc_component_driver {
+	const char *name;
+
+	...
+	int (*probe)(struct snd_soc_component *);
+	void (*remove)(struct snd_soc_component *);
+	int (*suspend)(struct snd_soc_component *);
+	int (*resume)(struct snd_soc_component *);
+
+	/* pcm creation and destruction */
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
+	void (*pcm_free)(struct snd_pcm *);
+
+	...
+	const struct snd_pcm_ops *ops;
+	const struct snd_compr_ops *compr_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.rst for a description of items 1 - 4.
+
+
+SoC DSP Drivers
+===============
+
+Each SoC DSP driver usually supplies the following features :-
+
+1. DAPM graph
+2. Mixer controls
+3. DMA IO to/from DSP buffers (if applicable)
+4. Definition of DSP front end (FE) PCM devices.
+
+Please see DPCM.txt for a description of item 4.
diff --git a/Documentation/sound/soc/pops-clicks.rst b/Documentation/sound/soc/pops-clicks.rst
new file mode 100644
index 000000000..de7eb2a66
--- /dev/null
+++ b/Documentation/sound/soc/pops-clicks.rst
@@ -0,0 +1,55 @@
+=====================
+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.