[ALSA] ASoC: documentation & maintainer

This patch adds documentation describing the ASoC architecture and a
maintainer entry for ASoC.
The documentation includes the following files:-
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.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>

Documentation/sound/alsa/soc/clocking.txt

+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.
+
+Bit Clock is usually always a ratio of MCLK or a multiple of LRC. i.e.
+
+BCLK = MCLK / x
+
+ or
+
+BCLK = LRC * x
+
+This relationship depends on the codec or SoC CPU in particular. ASoC can quite
+easily match a codec that generates BCLK by division (FSBD) with a CPU that
+generates BCLK by multiplication (FSB).
+
+
+ASoC Clocking
+-------------
+
+The ASoC core determines the clocking for each particular configuration at
+runtime. This is to allow for dynamic audio clocking wereby the audio clock is
+variable and depends on the system state or device usage scenario. i.e. a voice
+call requires slower clocks (and hence less power) than MP3 playback.
+
+ASoC will call the config_sysclock() function for the target machine during the
+audio parameters configuration. The function is responsible for then clocking
+the machine audio subsytem and returning the audio clock speed to the core.
+This function should also call the codec and cpu DAI clock_config() functions
+to configure their respective internal clocking if required.
+
+
+ASoC Clocking Control Flow
+--------------------------
+
+The ASoC core will call the machine drivers config_sysclock() when most of the
+DAI capabilities are known. The machine driver is then responsible for calling
+the codec and/or CPU DAI drivers with the selected capabilities and the current
+MCLK. Note that the machine driver is also resonsible for setting the MCLK (and
+enabling it).
+
+   (1) Match Codec and CPU DAI capabilities. At this point we have
+       matched the majority of the DAI fields and now need to make sure this
+       mode is currently clockable.
+
+   (2) machine->config_sysclk() is now called with the matched DAI FS, sample
+       rate and BCLK master. This function then gets/sets the current audio
+       clock (depening on usage) and calls the codec and CPUI DAI drivers with
+       the FS, rate, BCLK master and MCLK.
+
+   (3) Codec/CPU DAI config_sysclock(). This function checks that the FS, rate,
+       BCLK master and MCLK are acceptable for the codec or CPU DAI. It also
+       sets the DAI internal state to work with said clocks.
+
+The config_sysclk() functions for CPU, codec and machine should return the MCLK
+on success and 0 on failure.
+
+
+Examples (b = BCLK, l = LRC)
+============================
+
+Example 1
+---------
+
+Simple codec that only runs at 48k @ 256FS in master mode.
+
+CPU only runs as slave DAI, however it generates a variable MCLK.
+
+             --------                 ---------
+            |        | <----mclk---  |         |
+            | Codec  |b -----------> |  CPU    |
+            |        |l -----------> |         |
+            |        |               |         |
+             --------                 ---------
+
+The codec driver has the following config_sysclock()
+
+	static unsigned int config_sysclk(struct snd_soc_codec_dai *dai,
+		struct snd_soc_clock_info *info, unsigned int clk)
+	{
+		/* make sure clock is 256 * rate */
+		if(info->rate << 8 == clk) {
+			dai->mclk = clk;
+			return clk;
+		}
+
+		return 0;
+	}
+
+The CPU I2S DAI driver has the following config_sysclk()
+
+	static unsigned int config_sysclk(struct snd_soc_codec_dai *dai,
+		struct snd_soc_clock_info *info, unsigned int clk)
+	{
+		/* can we support this clk */
+		if(set_audio_clk(clk) < 0)
+			return -EINVAL;
+
+		dai->mclk = clk;
+		return dai->clk;
+	}
+
+The machine driver config_sysclk() in this example is as follows:-
+
+	unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd,
+		struct snd_soc_clock_info *info)
+	{
+		int clk = info->rate * info->fs;
+
+		/* check that CPU can deliver clock */
+		if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0)
+			return -EINVAL;
+
+		/* can codec work with this clock */
+		return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk);
+	}
+
+
+Example 2
+---------
+
+Codec that can master at 8k and 48k at various FS (and hence supports a fixed
+set of input MCLK's) and can also be slave at various FS .
+
+The CPU can master at 8k and 48k @256 FS and can be slave at any FS.
+
+MCLK is a 12.288MHz crystal on this machine.
+
+             --------                 ---------
+            |        |  <---xtal---> |         |
+            | Codec  |b <----------> |  CPU    |
+            |        |l <----------> |         |
+            |        |               |         |
+             --------                 ---------
+
+
+The codec driver has the following config_sysclock()
+
+	/* supported input clocks */
+	const static int hifi_clks[] = {11289600, 12000000, 12288000,
+		16934400, 18432000};
+
+	static unsigned int config_hsysclk(struct snd_soc_codec_dai *dai,
+		struct snd_soc_clock_info *info, unsigned int clk)
+	{
+		int i;
+
+		/* is clk supported  */
+		for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) {
+			if(clk == hifi_clks[i]) {
+				dai->mclk = clk;
+				return clk;
+			}
+		}
+
+		/* this clk is not supported */
+		return 0;
+	}
+
+The CPU I2S DAI driver has the following config_sysclk()
+
+	static unsigned int config_sysclk(struct snd_soc_codec_dai *dai,
+		struct snd_soc_clock_info *info, unsigned int clk)
+	{
+		/* are we master or slave */
+		if (info->bclk_master &
+			(SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+
+			/* we can only master @ 256FS */
+			if(info->rate << 8 == clk) {
+				dai->mclk = clk;
+				return dai->mclk;
+			}
+		} else {
+			/* slave we can run at any FS */
+			dai->mclk = clk;
+			return dai->mclk;
+		}
+
+		/* not supported */
+		return dai->clk;
+	}
+
+The machine driver config_sysclk() in this example is as follows:-
+
+	unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd,
+		struct snd_soc_clock_info *info)
+	{
+		int clk = 12288000; /* 12.288MHz */
+
+		/* who's driving the link */
+		if (info->bclk_master &
+			(SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+			/* codec master */
+
+			/* check that CPU can work with clock */
+			if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0)
+				return -EINVAL;
+
+			/* can codec work with this clock */
+			return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk);
+		} else {
+			/* cpu master */
+
+			/* check that codec can work with clock */
+			if(rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk) < 0)
+				return -EINVAL;
+
+			/* can CPU work with this clock */
+			return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk);
+		}
+	}
+
+
+
+Example 3
+---------
+
+Codec that masters at 8k ... 48k @256 FS. Codec can also be slave and
+doesn't care about FS. The codec has an internal PLL and dividers to generate
+the necessary internal clocks (for 256FS).
+
+CPU can only be slave and doesn't care about FS.
+
+MCLK is a non controllable 13MHz clock from the CPU.
+
+
+             --------                 ---------
+            |        | <----mclk---  |         |
+            | Codec  |b <----------> |  CPU    |
+            |        |l <----------> |         |
+            |        |               |         |
+             --------                 ---------
+
+The codec driver has the following config_sysclock()
+
+	/* valid PCM clock dividers * 2 */
+	static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16};
+
+	static unsigned int config_vsysclk(struct snd_soc_codec_dai *dai,
+		struct snd_soc_clock_info *info, unsigned int clk)
+	{
+		int i, j, best_clk = info->fs * info->rate;
+
+		/* can we run at this clk without the PLL ? */
+		for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) {
+			if ((best_clk >> 1) * pcm_divs[i] == clk) {
+				dai->pll_in = 0;
+				dai->clk_div = pcm_divs[i];
+				dai->mclk = best_clk;
+				return dai->mclk;
+			}
+		}
+
+		/* now check for PLL support */
+		for (i = 0; i < ARRAY_SIZE(pll_div); i++) {
+			if (pll_div[i].pll_in == clk) {
+				for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) {
+					if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) {
+						dai->pll_in = clk;
+						dai->pll_out = pll_div[i].pll_out;
+						dai->clk_div = pcm_divs[j];
+						dai->mclk = best_clk;
+						return dai->mclk;
+					}
+				}
+			}
+		}
+
+		/* this clk is not supported */
+		return 0;
+	}
+
+
+The CPU I2S DAI driver has the does not need a config_sysclk() as it can slave
+at any FS.
+
+	unsigned int config_sysclk(struct snd_soc_pcm_runtime *rtd,
+		struct snd_soc_clock_info *info)
+	{
+		/* codec has pll that generates mclk from 13MHz xtal */
+		return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000);
+	}

Documentation/sound/alsa/soc/codec.txt

+ASoC Codec Driver
+=================
+
+The codec driver is generic and hardware independent code that configures the
+codec to provide audio capture and playback. It should contain no code that is
+specific to the target platform or machine. All platform and machine specific
+code should be added to the platform and machine drivers respectively.
+
+Each codec driver must provide the following features:-
+
+ 1) Digital audio interface (DAI) description
+ 2) Digital audio interface configuration
+ 3) PCM's description
+ 4) Codec control IO - using I2C, 3 Wire(SPI) or both API's
+ 5) Mixers and audio controls
+ 6) Sysclk configuration
+ 7) Codec audio operations
+
+Optionally, codec drivers can also provide:-
+
+ 8) DAPM description.
+ 9) DAPM event handler.
+10) DAC Digital mute control.
+
+It's probably best to use this guide in conjuction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+1 - Digital Audio Interface (DAI) description
+---------------------------------------------
+The DAI is a digital audio data transfer link between the codec and host SoC
+CPU. It typically has data transfer capabilities in both directions
+(playback and capture) and can run at a variety of different speeds.
+Supported interfaces currently include AC97, I2S and generic PCM style links.
+Please read DAI.txt for implementation information.
+
+
+2 - Digital Audio Interface (DAI) configuration
+-----------------------------------------------
+DAI configuration is handled by the codec_pcm_prepare function and is
+responsible for configuring and starting the DAI on the codec. This can be
+called multiple times and is atomic. It can access the runtime parameters.
+
+This usually consists of a large function with numerous switch statements to
+set up each configuration option. These options are set by the core at runtime.
+
+
+3 - Codec PCM's
+---------------
+Each codec must have it's PCM's defined. This defines the number of channels,
+stream names, callbacks and codec name. It is also used to register the DAI
+with the ASoC core. The PCM structure also associates the DAI capabilities with
+the ALSA PCM.
+
+e.g.
+
+static struct snd_soc_pcm_codec wm8731_pcm_client = {
+	.name = "WM8731",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+	},
+	.config_sysclk = wm8731_config_sysclk,
+	.ops = {
+		.prepare = wm8731_pcm_prepare,
+	},
+	.caps = {
+		.num_modes = ARRAY_SIZE(wm8731_hwfmt),
+		.modes = &wm8731_hwfmt[0],
+	},
+};
+
+
+4 - Codec control IO
+--------------------
+The codec can ususally be controlled via an I2C or SPI style interface (AC97
+combines control with data in the DAI). The codec drivers will have to provide
+functions to read and write the codec registers along with supplying a register
+cache:-
+
+	/* IO control data and register cache */
+    void *control_data; /* codec control (i2c/3wire) data */
+    void *reg_cache;
+
+Codec read/write should do any data formatting and call the hardware read write
+below to perform the IO. These functions are called by the core and alsa when
+performing DAPM or changing the mixer:-
+
+    unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+    int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
+
+Codec hardware IO functions - usually points to either the I2C, SPI or AC97
+read/write:-
+
+	hw_write_t hw_write;
+	hw_read_t hw_read;
+
+
+5 - Mixers and audio controls
+-----------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+
+    #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+
+  xname = Control name e.g. "Playback Volume"
+  reg = codec register
+  shift = control bit(s) offset in register
+  mask = control bit size(s) e.g. mask of 7 = 3 bits
+  invert = the control is inverted
+
+Other macros include:-
+
+    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+
+    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+
+    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+
+   xreg = register
+   xshift = control bit(s) offset in register
+   xmask = control bit(s) size
+   xtexts = pointer to array of strings that describe each setting
+
+   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+6 - System clock configuration.
+-------------------------------
+The system clock that drives the audio subsystem can change depending on sample
+rate and the system power state. i.e.
+
+o Higher sample rates sometimes need a higher system clock.
+o Low system power states can sometimes limit the available clocks.
+
+This function is a callback that the machine driver can call to set and
+determine if the clock and sample rate combination is supported by the codec at
+the present time (and system state).
+
+NOTE: If the codec has a PLL then it has a lot more flexability wrt clock and
+sample rate combinations.
+
+Your config_sysclock function should return the MCLK if it's a valid
+combination for your codec else 0;
+
+Please read clocking.txt now.
+
+
+7 - Codec Audio Operations
+--------------------------
+The codec driver also supports the following alsa operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(snd_pcm_substream_t *);
+	void (*shutdown)(snd_pcm_substream_t *);
+	int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *);
+	int (*hw_free)(snd_pcm_substream_t *);
+	int (*prepare)(snd_pcm_substream_t *);
+};
+
+Please refer to the alsa driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
+
+
+8 - DAPM description.
+---------------------
+The Dynamic Audio Power Management description describes the codec's power
+components, their relationships and registers to the ASoC core. Please read
+dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+9 - DAPM event handler
+----------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep
+when not in use.
+
+Power states:-
+
+	SNDRV_CTL_POWER_D0: /* full On */
+	/* vref/mid, clk and osc on, active */
+
+	SNDRV_CTL_POWER_D1: /* partial On */
+	SNDRV_CTL_POWER_D2: /* partial On */
+
+	SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	/* everything off except vref/vmid, inactive */
+
+	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+10 - Codec DAC digital mute control.
+------------------------------------
+Most codecs have a digital mute before the DAC's that can be used to minimise
+any system noise.  The mute stops any digital data from entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI when the
+mute is applied or freed.
+
+i.e.
+
+static int wm8974_mute(struct snd_soc_codec *codec,
+	struct snd_soc_codec_dai *dai, int mute)
+{
+	u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
+	if(mute)
+		wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
+	else
+		wm8974_write(codec, WM8974_DAC, mute_reg);
+	return 0;
+}

Documentation/sound/alsa/soc/machine.txt

+ASoC Machine Driver
+===================
+
+The ASoC machine (or board) driver is the code that glues together the platform
+and codec drivers.
+
+The machine driver can contain codec and platform specific code. It registers
+the audio subsystem with the kernel as a platform device and is represented by
+the following struct:-
+
+/* SoC machine */
+struct snd_soc_machine {
+	char *name;
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+
+	/* the pre and post PM functions are used to do any PM work before and
+	 * after the codec and DAI's do any PM work. */
+	int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
+	int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
+	int (*resume_pre)(struct platform_device *pdev);
+	int (*resume_post)(struct platform_device *pdev);
+
+	/* machine stream operations */
+	struct snd_soc_ops *ops;
+
+	/* CPU <--> Codec DAI links  */
+	struct snd_soc_dai_link *dai_link;
+	int num_links;
+};
+
+probe()/remove()
+----------------
+probe/remove are optional. Do any machine specific probe here.
+
+
+suspend()/resume()
+------------------
+The machine driver has pre and post versions of suspend and resume to take care
+of any machine audio tasks that have to be done before or after the codec, DAI's
+and DMA is suspended and resumed. Optional.
+
+
+Machine operations
+------------------
+The machine specific audio operations can be set here. Again this is optional.
+
+
+Machine DAI Configuration
+-------------------------
+The machine DAI configuration glues all the codec and CPU DAI's together. It can
+also be used to set up the DAI system clock and for any machine related DAI
+initialisation e.g. the machine audio map can be connected to the codec audio
+map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c
+for examples.
+
+struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &wm8731_dai,
+	.init = corgi_wm8731_init,
+	.config_sysclk = corgi_config_sysclk,
+};
+
+struct snd_soc_machine then sets up the machine with it's DAI's. e.g.
+
+/* corgi audio machine driver */
+static struct snd_soc_machine snd_soc_machine_corgi = {
+	.name = "Corgi",
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+	.ops = &corgi_ops,
+};
+
+
+Machine Audio Subsystem
+-----------------------
+
+The machine soc device glues the platform, machine and codec driver together.
+Private data can also be set here. e.g.
+
+/* corgi audio private data */
+static struct wm8731_setup_data corgi_wm8731_setup = {
+	.i2c_address = 0x1b,
+};
+
+/* corgi audio subsystem */
+static struct snd_soc_device corgi_snd_devdata = {
+	.machine = &snd_soc_machine_corgi,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &corgi_wm8731_setup,
+};
+
+
+Machine Power Map
+-----------------
+
+The machine driver can optionally extend the codec power map and to become an
+audio power map of the audio subsystem. This allows for automatic power up/down
+of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
+sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
+details.
+
+
+Machine Controls
+----------------
+
+Machine specific audio mixer controls can be added in the dai init function.
\ No newline at end of file

Documentation/sound/alsa/soc/overview.txt

+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

Documentation/sound/alsa/soc/platform.txt

+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.

Documentation/sound/alsa/soc/pops_clicks.txt

+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.

MAINTAINERS

@@ -3037,6 +3037,12 @@ M:	perex@suse.cz
 L:	alsa-devel@alsa-project.org
 S:	Maintained
 
+SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
+P:	Liam Girdwood
+M:	liam.girdwood@wolfsonmicro.com
+L:	alsa-devel@alsa-project.org
+S:	Supported
+
 SPI SUBSYSTEM
 P:	David Brownell
 M:	dbrownell@users.sourceforge.net