[PATCH 00/10] ASoC: Modernize documents

[PATCH 00/10] ASoC: Modernize documents

[Takashi Iwai]

Hi,

this patchset converts the ASoC text documents to ReST format, which
has been recently introduced to the kernel tree.  The conversions are
really simple since the original texts were already well formatted.

The files are moved to Documentation/sound/soc, and TOC is to the
index page.

All the patches are found in topic/rest-docs branch of sound git
tree.

The new file after 4.9 isn't covered yet, as the branch is based on
4.9-rc4.  I found only soc/codec_to_codec.txt in linux-next, and it
seems easy to convert.  Mark, let me know how you'd like to organize.
I can pull your branch and convert in my side, or you pull mine and do
it on yours.


thanks,

Takashi

===

Takashi Iwai (10):
  ASoC: doc: ReSTize overview.txt
  ASoC: doc: ReSTize codec.txt
  ASoC: doc: ReSTize DAI.txt
  ASoC: doc: ReSTize dapm.txt
  ASoC: doc: ReSTize platform.txt
  ASoC: doc: ReSTize machine.txt
  ASoC: doc: ReSTize pops_clicks.txt
  ASoC: doc: ReSTize clocking.txt
  ASoC: doc: ReSTize jack.txt
  ASoC: doc: ReSTize DPCM.txt

 Documentation/sound/index.rst                      |   1 +
 .../{alsa/soc/clocking.txt => soc/clocking.rst}    |  13 +-
 .../sound/{alsa/soc/codec.txt => soc/codec.rst}    |  75 +++---
 .../sound/{alsa/soc/DAI.txt => soc/dai.rst}        |  28 ++-
 .../sound/{alsa/soc/dapm.txt => soc/dapm.rst}      | 249 ++++++++++--------
 .../sound/{alsa/soc/DPCM.txt => soc/dpcm.rst}      | 280 +++++++++++----------
 Documentation/sound/soc/index.rst                  |  19 ++
 .../sound/{alsa/soc/jack.txt => soc/jack.rst}      |   1 +
 .../{alsa/soc/machine.txt => soc/machine.rst}      |  22 +-
 .../{alsa/soc/overview.txt => soc/overview.rst}    |  32 +--
 .../{alsa/soc/platform.txt => soc/platform.rst}    |  31 +--
 .../soc/pops_clicks.txt => soc/pops-clicks.rst}    |  15 +-
 12 files changed, 417 insertions(+), 349 deletions(-)
 rename Documentation/sound/{alsa/soc/clocking.txt => soc/clocking.rst} (94%)
 rename Documentation/sound/{alsa/soc/codec.txt => soc/codec.rst} (84%)
 rename Documentation/sound/{alsa/soc/DAI.txt => soc/dai.rst} (71%)
 rename Documentation/sound/{alsa/soc/dapm.txt => soc/dapm.rst} (62%)
 rename Documentation/sound/{alsa/soc/DPCM.txt => soc/dpcm.rst} (57%)
 create mode 100644 Documentation/sound/soc/index.rst
 rename Documentation/sound/{alsa/soc/jack.txt => soc/jack.rst} (99%)
 rename Documentation/sound/{alsa/soc/machine.txt => soc/machine.rst} (90%)
 rename Documentation/sound/{alsa/soc/overview.txt => soc/overview.rst} (83%)
 rename Documentation/sound/{alsa/soc/platform.txt => soc/platform.rst} (81%)
 rename Documentation/sound/{alsa/soc/pops_clicks.txt => soc/pops-clicks.rst} (87%)

-- 
2.10.2

[PATCH 01/10] ASoC: doc: ReSTize overview.txt

[Takashi Iwai]

A simple conversion from a plain text file.
Created a new subdirectory, Documentation/sound/soc, for this and
other ASoC documents.

Since the index page contains the TOC, so "Documentation" section got
removed from overview.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/index.rst                      |  1 +
 Documentation/sound/soc/index.rst                  | 10 +++++++
 .../{alsa/soc/overview.txt => soc/overview.rst}    | 32 ++--------------------
 3 files changed, 14 insertions(+), 29 deletions(-)
 create mode 100644 Documentation/sound/soc/index.rst
 rename Documentation/sound/{alsa/soc/overview.txt => soc/overview.rst} (83%)

diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst
index 1f5d166f81c4..47b89f014e69 100644
--- a/Documentation/sound/index.rst
+++ b/Documentation/sound/index.rst
@@ -7,6 +7,7 @@ Linux Sound Subsystem Documentation
 
    kernel-api/index
    designs/index
+   soc/index
    alsa-configuration
    hd-audio/index
    cards/index
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
new file mode 100644
index 000000000000..e974fd9f38a3
--- /dev/null
+++ b/Documentation/sound/soc/index.rst
@@ -0,0 +1,10 @@
+==============
+ALSA SoC Layer
+==============
+
+The documentation is spilt into the following sections:-
+
+.. toctree::
+   :maxdepth: 2
+
+   overview
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/soc/overview.rst
similarity index 83%
rename from Documentation/sound/alsa/soc/overview.txt
rename to Documentation/sound/soc/overview.rst
index f3f28b7ae242..dc8370bbfff6 100644
--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/soc/overview.rst
@@ -1,5 +1,6 @@
-ALSA SoC Layer
-==============
+=======================
+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.
@@ -66,30 +67,3 @@ multiple re-usable component drivers :-
     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).
-
-
-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.
-
-jack.txt: ASoC jack detection.
-
-DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.
-- 
2.10.2

[PATCH 02/10] ASoC: doc: ReSTize codec.txt

[Takashi Iwai]

A simple conversion from a plain text file.
The section numbers are dropped to align with other documents.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 .../sound/{alsa/soc/codec.txt => soc/codec.rst}    | 75 +++++++++++++---------
 Documentation/sound/soc/index.rst                  |  1 +
 2 files changed, 44 insertions(+), 32 deletions(-)
 rename Documentation/sound/{alsa/soc/codec.txt => soc/codec.rst} (84%)

diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/soc/codec.rst
similarity index 84%
rename from Documentation/sound/alsa/soc/codec.txt
rename to Documentation/sound/soc/codec.rst
index db5f9c9ae149..f87612b94812 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/soc/codec.rst
@@ -1,3 +1,4 @@
+=======================
 ASoC Codec Class Driver
 =======================
 
@@ -9,16 +10,16 @@ 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.
+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.
+7. DAC Digital mute control.
 
 Its probably best to use this guide in conjunction with the existing codec
 driver code in sound/soc/codecs/
@@ -26,24 +27,25 @@ driver code in sound/soc/codecs/
 ASoC Codec driver breakdown
 ===========================
 
-1 - Codec DAI and PCM configuration
------------------------------------
+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 = {
+  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 = {
+  };
+  
+  struct snd_soc_dai_driver wm8731_dai = {
 	.name = "wm8731-hifi",
 	.playback = {
 		.stream_name = "Playback",
@@ -59,25 +61,27 @@ struct snd_soc_dai_driver wm8731_dai = {
 		.formats = WM8731_FORMATS,},
 	.ops = &wm8731_dai_ops,
 	.symmetric_rates = 1,
-};
+  };
 
 
-2 - Codec control IO
---------------------
+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.
 
 
-3 - Mixers and audio controls
------------------------------
+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
@@ -86,18 +90,22 @@ Defines a single control as follows:-
   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
@@ -109,25 +117,26 @@ Defines an single enumerated control as follows:-
 Defines a stereo enumerated control
 
 
-4 - Codec Audio Operations
---------------------------
+Codec Audio Operations
+----------------------
 The codec driver also supports the following ALSA PCM operations:-
+::
 
-/* SoC audio ops */
-struct snd_soc_ops {
+  /* 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.
----------------------
+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.
@@ -135,13 +144,14 @@ Please read dapm.txt for details of building the description.
 Please also see the examples in other codec drivers.
 
 
-6 - DAPM event handler
-----------------------
+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 */
@@ -155,8 +165,8 @@ Power states:-
 	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
 
 
-7 - Codec DAC digital mute control
-----------------------------------
+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.
@@ -165,9 +175,10 @@ 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)
-{
+  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;
 
@@ -176,4 +187,4 @@ static int wm8974_mute(struct snd_soc_dai *dai, int mute)
 	else
 		snd_soc_write(codec, WM8974_DAC, mute_reg);
 	return 0;
-}
+  }
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index e974fd9f38a3..a2e023c91df2 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -8,3 +8,4 @@ The documentation is spilt into the following sections:-
    :maxdepth: 2
 
    overview
+   codec
-- 
2.10.2

[PATCH 03/10] ASoC: doc: ReSTize DAI.txt

[Takashi Iwai]

A simple conversion from a plain text file with slight reformatting /
corrections.

The file name was changed to lower letters to align with others.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 .../sound/{alsa/soc/DAI.txt => soc/dai.rst}        | 28 ++++++++++++++--------
 Documentation/sound/soc/index.rst                  |  1 +
 2 files changed, 19 insertions(+), 10 deletions(-)
 rename Documentation/sound/{alsa/soc/DAI.txt => soc/dai.rst} (71%)

diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/soc/dai.rst
similarity index 71%
rename from Documentation/sound/alsa/soc/DAI.txt
rename to Documentation/sound/soc/dai.rst
index c9679264c559..55820e51708f 100644
--- a/Documentation/sound/alsa/soc/DAI.txt
+++ b/Documentation/sound/soc/dai.rst
@@ -1,3 +1,7 @@
+==================================
+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.
 
@@ -5,21 +9,21 @@ 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
+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 :-
+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
+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
@@ -30,13 +34,15 @@ 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.
+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.
+Left Justified
+  MSB is transmitted on transition of LRC.
 
- o Right Justified - MSB is transmitted sample size BCLKs before LRC
-                     transition.
+Right Justified
+  MSB is transmitted sample size BCLKs before LRC transition.
 
 PCM
 ===
@@ -51,6 +57,8 @@ 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.
+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.
+Mode B
+  MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index a2e023c91df2..aea7ae7e5aad 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -9,3 +9,4 @@ The documentation is spilt into the following sections:-
 
    overview
    codec
+   dai
-- 
2.10.2

[PATCH 04/10] ASoC: doc: ReSTize dapm.txt

[Takashi Iwai]

A simple conversion from a plain text file.
The section numbers and the item numbers are dropped to align with the
ReST format.  Some lists are converted to description lists to be
clearer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 .../sound/{alsa/soc/dapm.txt => soc/dapm.rst}      | 249 ++++++++++++---------
 Documentation/sound/soc/index.rst                  |   1 +
 2 files changed, 144 insertions(+), 106 deletions(-)
 rename Documentation/sound/{alsa/soc/dapm.txt => soc/dapm.rst} (62%)

diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/soc/dapm.rst
similarity index 62%
rename from Documentation/sound/alsa/soc/dapm.txt
rename to Documentation/sound/soc/dapm.rst
index c45bd79f291e..a27f42befa4d 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/soc/dapm.rst
@@ -1,8 +1,9 @@
+===================================================
 Dynamic Audio Power Management for Portable Devices
 ===================================================
 
-1. Description
-==============
+Description
+===========
 
 Dynamic Audio Power Management (DAPM) is designed to allow portable
 Linux devices to use the minimum amount of power within the audio
@@ -21,20 +22,28 @@ level power systems.
 
 There are 4 power domains within DAPM
 
-   1. Codec bias domain - VREF, VMID (core codec and audio power)
+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.
 
-   2. Platform/Machine domain - physically connected inputs and outputs
+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 subsystem signal paths
+Path domain
+      audio subsystem signal paths
+
       Automatically set when mixer and mux settings are changed by the user.
       e.g. alsamixer, amixer.
 
-   4. Stream domain - DACs and ADCs.
+Stream domain
+      DACs and ADCs.
+
       Enabled and disabled when stream playback/capture is started and
       stopped respectively. e.g. aplay, arecord.
 
@@ -45,34 +54,57 @@ internal codec components). All audio components that effect power are called
 widgets hereafter.
 
 
-2. DAPM Widgets
-===============
+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 Regulator  - External regulator that supplies power to audio components.
- o Clock      -	External clock that supplies clock to audio components.
- o AIF IN     - Audio Interface Input (with TDM slot mask).
- o AIF OUT    - Audio Interface Output (with TDM slot mask).
- o Siggen     - Signal Generator.
- o DAI IN     - Digital Audio Interface Input.
- o DAI OUT    - Digital Audio Interface Output.
- o DAI Link   - DAI Link between two DAI structures */
- o Pre        - Special PRE widget (exec before all others)
- o Post       - Special POST widget (exec after all others)
+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)
 
 (Widgets are defined in include/sound/soc-dapm.h)
 
@@ -84,52 +116,57 @@ 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 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)
+  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),
+  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),
+  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),
 
 
-2.2 Path Domain Widgets
------------------------
+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)
+  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),
-};
+  /* 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,
+  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,
@@ -137,48 +174,49 @@ you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
 as for SND_SOC_DAPM_MIXER.
 
 
-2.3 Machine domain Widgets
---------------------------
+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.
 
- o Speaker Amp
- o Microphone Bias
- o Jack connectors
+* 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:-
+when the Mic is inserted:-::
 
-static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
-{
+  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),
+  SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
 
 
-2.4 Codec (BIAS) Domain
------------------------
+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.
 
 
-2.5 Virtual Widgets
--------------------
+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),
+  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.
 
@@ -186,8 +224,8 @@ 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/DSP Widget Interconnections
-====================================
+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
@@ -201,13 +239,14 @@ 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
+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"},
@@ -216,22 +255,17 @@ connect the destination widget (wrt audio signal) with its source widgets.
 
 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".
+* 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);
+  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
@@ -239,12 +273,13 @@ 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
+-------------------------------
 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"},
@@ -254,52 +289,54 @@ 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
-===================
+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
+* 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.
 
 
-5 DAPM Widget Events
-====================
+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)
-{
+  /* 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 =
+  /* 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
----------------
+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 */
+::
+
+  /* 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/index.rst b/Documentation/sound/soc/index.rst
index aea7ae7e5aad..0c0c38e582b4 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -10,3 +10,4 @@ The documentation is spilt into the following sections:-
    overview
    codec
    dai
+   dapm
-- 
2.10.2

[PATCH 05/10] ASoC: doc: ReSTize platform.txt

[Takashi Iwai]

A simple conversion from a plain text file.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/soc/index.rst                  |  1 +
 .../{alsa/soc/platform.txt => soc/platform.rst}    | 31 ++++++++++++----------
 2 files changed, 18 insertions(+), 14 deletions(-)
 rename Documentation/sound/{alsa/soc/platform.txt => soc/platform.rst} (81%)

diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 0c0c38e582b4..c5a55195bf4d 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -11,3 +11,4 @@ The documentation is spilt into the following sections:-
    codec
    dai
    dapm
+   platform
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/soc/platform.rst
similarity index 81%
rename from Documentation/sound/alsa/soc/platform.txt
rename to Documentation/sound/soc/platform.rst
index 3a08a2c9150c..d5574904d981 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/soc/platform.rst
@@ -1,3 +1,4 @@
+====================
 ASoC Platform Driver
 ====================
 
@@ -9,21 +10,23 @@ Audio DMA
 =========
 
 The platform DMA driver optionally supports the following ALSA operations:-
+::
 
-/* SoC audio ops */
-struct snd_soc_ops {
+  /* 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 {
+  struct snd_soc_platform_driver {
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -44,7 +47,7 @@ struct snd_soc_platform_driver {
 
 	/* 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/
@@ -57,11 +60,11 @@ 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)
+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.
 
@@ -71,9 +74,9 @@ 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.
+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.
-- 
2.10.2

[PATCH 06/10] ASoC: doc: ReSTize machine.txt

[Takashi Iwai]

A simple conversion from a plain text file.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/soc/index.rst                  |  1 +
 .../{alsa/soc/machine.txt => soc/machine.rst}      | 22 +++++++++++++---------
 2 files changed, 14 insertions(+), 9 deletions(-)
 rename Documentation/sound/{alsa/soc/machine.txt => soc/machine.rst} (90%)

diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index c5a55195bf4d..4ac3585e7dd1 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -12,3 +12,4 @@ The documentation is spilt into the following sections:-
    dai
    dapm
    platform
+   machine
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/soc/machine.rst
similarity index 90%
rename from Documentation/sound/alsa/soc/machine.txt
rename to Documentation/sound/soc/machine.rst
index 6bf2d2063b52..515c9444deaf 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/soc/machine.rst
@@ -1,3 +1,4 @@
+===================
 ASoC Machine Driver
 ===================
 
@@ -9,9 +10,10 @@ 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 {
+  /* SoC machine */
+  struct snd_soc_card {
 	char *name;
 
 	...
@@ -33,7 +35,7 @@ struct snd_soc_card {
 	int num_links;
 
 	...
-};
+  };
 
 probe()/remove()
 ----------------
@@ -55,9 +57,10 @@ 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 = {
+  /* 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",
@@ -66,16 +69,17 @@ static struct snd_soc_dai_link corgi_dai = {
 	.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 = {
+  /* corgi audio machine driver */
+  static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
 	.dai_link = &corgi_dai,
 	.num_links = 1,
-};
+  };
 
 
 Machine Power Map
-- 
2.10.2

[PATCH 07/10] ASoC: doc: ReSTize pops_clicks.txt

[Takashi Iwai]

A simple conversion from a plain text file.
The file name was changed from "pops_clicks" to "pops-clicks" to align
with others.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/soc/index.rst                         |  1 +
 .../{alsa/soc/pops_clicks.txt => soc/pops-clicks.rst}     | 15 +++++++++------
 2 files changed, 10 insertions(+), 6 deletions(-)
 rename Documentation/sound/{alsa/soc/pops_clicks.txt => soc/pops-clicks.rst} (87%)

diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 4ac3585e7dd1..4dd679927a91 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -13,3 +13,4 @@ The documentation is spilt into the following sections:-
    dapm
    platform
    machine
+   pops-clicks
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/soc/pops-clicks.rst
similarity index 87%
rename from Documentation/sound/alsa/soc/pops_clicks.txt
rename to Documentation/sound/soc/pops-clicks.rst
index e1e74daa4497..de7eb2a6604a 100644
--- a/Documentation/sound/alsa/soc/pops_clicks.txt
+++ b/Documentation/sound/soc/pops-clicks.rst
@@ -1,3 +1,4 @@
+=====================
 Audio Pops and Clicks
 =====================
 
@@ -20,10 +21,11 @@ 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
+   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.
@@ -36,10 +38,11 @@ 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
+   Startup Order - Input PGA --> Mixers --> ADC
+  
+   Shutdown Order - ADC --> Mixers --> Input PGA
 
 
 Zipper Noise
-- 
2.10.2

[PATCH 08/10] ASoC: doc: ReSTize clocking.txt

[Takashi Iwai]

A simple conversion from a plain text file.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 .../sound/{alsa/soc/clocking.txt => soc/clocking.rst}       | 13 ++++---------
 Documentation/sound/soc/index.rst                           |  1 +
 2 files changed, 5 insertions(+), 9 deletions(-)
 rename Documentation/sound/{alsa/soc/clocking.txt => soc/clocking.rst} (94%)

diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/soc/clocking.rst
similarity index 94%
rename from Documentation/sound/alsa/soc/clocking.txt
rename to Documentation/sound/soc/clocking.rst
index b1300162e01c..32122d6877a3 100644
--- a/Documentation/sound/alsa/soc/clocking.txt
+++ b/Documentation/sound/soc/clocking.rst
@@ -1,3 +1,4 @@
+==============
 Audio Clocking
 ==============
 
@@ -30,15 +31,9 @@ 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
+- 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
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 4dd679927a91..0055abe16e7e 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -14,3 +14,4 @@ The documentation is spilt into the following sections:-
    platform
    machine
    pops-clicks
+   clocking
-- 
2.10.2

[PATCH 09/10] ASoC: doc: ReSTize jack.txt

[Takashi Iwai]

A simple conversion from a plain text file.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/soc/index.rst                       | 1 +
 Documentation/sound/{alsa/soc/jack.txt => soc/jack.rst} | 1 +
 2 files changed, 2 insertions(+)
 rename Documentation/sound/{alsa/soc/jack.txt => soc/jack.rst} (99%)

diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 0055abe16e7e..85ec51764e83 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -15,3 +15,4 @@ The documentation is spilt into the following sections:-
    machine
    pops-clicks
    clocking
+   jack
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/soc/jack.rst
similarity index 99%
rename from Documentation/sound/alsa/soc/jack.txt
rename to Documentation/sound/soc/jack.rst
index fcf82a417293..644b99ecba35 100644
--- a/Documentation/sound/alsa/soc/jack.txt
+++ b/Documentation/sound/soc/jack.rst
@@ -1,3 +1,4 @@
+===================
 ASoC jack detection
 ===================
 
-- 
2.10.2

[PATCH 10/10] ASoC: doc: ReSTize DPCM.txt

[Takashi Iwai]

A simple conversion from a plain text file.
The file name was renamed to lower letters to align with others.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 .../sound/{alsa/soc/DPCM.txt => soc/dpcm.rst}      | 280 +++++++++++----------
 Documentation/sound/soc/index.rst                  |   1 +
 2 files changed, 147 insertions(+), 134 deletions(-)
 rename Documentation/sound/{alsa/soc/DPCM.txt => soc/dpcm.rst} (57%)

diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/soc/dpcm.rst
similarity index 57%
rename from Documentation/sound/alsa/soc/DPCM.txt
rename to Documentation/sound/soc/dpcm.rst
index 0110180b7ac6..395e5a516282 100644
--- a/Documentation/sound/alsa/soc/DPCM.txt
+++ b/Documentation/sound/soc/dpcm.rst
@@ -1,8 +1,9 @@
+===========
 Dynamic PCM
 ===========
 
-1. Description
-==============
+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
@@ -23,22 +24,23 @@ 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
-                    *************
+::
+
+  | 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
@@ -55,50 +57,52 @@ 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
-                    *************
+::
+
+                      *************
+  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
-                    *************
+::
+
+                      *************
+  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.
+1. Machine driver receives Jack removal event.
 
- 2) Machine driver OR audio HAL disables the Headset path.
+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.
+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.
+4. Machine driver or audio HAL enables the speaker path.
 
- 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
-    trigger(start) for DAI1 Speakers since the path is enabled.
+5. DPCM runs the PCM ops for startup(), hw_params(), prepapre() 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
@@ -112,36 +116,38 @@ 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.
+1. Define the FE and BE DAI links.
 
- 2) Define any FE/BE PCM operations.
+2. Define any FE/BE PCM operations.
 
- 3) Define widget graph connections.
+3. Define widget graph connections.
 
 
-1 FE and BE DAI links
----------------------
+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
-                    *************
+  | 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[] = {
+  static struct snd_soc_dai_link machine_dais[] = {
 	{
 		.name = "PCM0 System",
 		.stream_name = "System Playback",
@@ -154,11 +160,11 @@ static struct snd_soc_dai_link machine_dais[] = {
 		.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"
+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
@@ -168,8 +174,9 @@ 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[] = {
+  static struct snd_soc_dai_link machine_dais[] = {
 	.....< FE DAI links here >
 	{
 		.name = "Codec Headset",
@@ -186,29 +193,30 @@ static struct snd_soc_dai_link machine_dais[] = {
 		.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 ``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
-                    *************
+::
+
+                      *************
+  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 whilst the DSP, MODEM DAI and the BT DAI are
 still in operation.
@@ -220,10 +228,10 @@ Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
 DSP firmware.
 
 
-2 FE/BE PCM operations
-----------------------
+FE/BE PCM operations
+--------------------
 
-The BE above also exports some PCM operations and a "fixup" callback. The fixup
+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.
 
@@ -231,10 +239,11 @@ e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo fo
 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,
+  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,
@@ -249,21 +258,22 @@ static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
 				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
 				    SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
-}
+  }
 
 The other PCM operation are the same as for regular DAI links. Use as necessary.
 
 
-3 Widget graph connections
---------------------------
+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"},
+  /* 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
@@ -273,24 +283,25 @@ 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.
+1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
 
- 2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
+2. DAPM graph showing DSP audio routing from FE DAIs to BEs.
 
- 3) DAPM widgets from DSP graph.
+3. DAPM widgets from DSP graph.
 
- 4) Mixers for gains, routing, etc.
+4. Mixers for gains, routing, etc.
 
- 5) DMA configuration.
+5. DMA configuration.
 
- 6) BE AIF widgets.
+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),
+  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).
@@ -301,33 +312,33 @@ 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
-                    *************
+::
+
+                      *************
+  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
+ 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
+ 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
@@ -339,16 +350,17 @@ 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 = {
+  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[] = {
+  static struct snd_soc_dai_link dais[] = {
 	< ... more DAI links above ... >
 	{
 		.name = "MODEM",
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 85ec51764e83..e142a0f25c5b 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -16,3 +16,4 @@ The documentation is spilt into the following sections:-
    pops-clicks
    clocking
    jack
+   dpcm
-- 
2.10.2

Re: [PATCH 00/10] ASoC: Modernize documents

[Mark Brown]

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On Thu, Nov 10, 2016 at 10:47:12PM +0100, Takashi Iwai wrote:

> The new file after 4.9 isn't covered yet, as the branch is based on
> 4.9-rc4.  I found only soc/codec_to_codec.txt in linux-next, and it
> seems easy to convert.  Mark, let me know how you'd like to organize.
> I can pull your branch and convert in my side, or you pull mine and do
> it on yours.

If you want to pull my branch in that's fine.

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Re: [PATCH 00/10] ASoC: Modernize documents

[Takashi Iwai]

On Fri, 11 Nov 2016 16:16:39 +0100,
Mark Brown wrote:
> 
> On Thu, Nov 10, 2016 at 10:47:12PM +0100, Takashi Iwai wrote:
> 
> > The new file after 4.9 isn't covered yet, as the branch is based on
> > 4.9-rc4.  I found only soc/codec_to_codec.txt in linux-next, and it
> > seems easy to convert.  Mark, let me know how you'd like to organize.
> > I can pull your branch and convert in my side, or you pull mine and do
> > it on yours.
> 
> If you want to pull my branch in that's fine.

OK, pulled now and worked on it.


Takashi