[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>
diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
new file mode 100644
index 0000000..919de76
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DAI.txt
@@ -0,0 +1,380 @@
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM.
+
+
+AC97
+====
+
+ AC97 is a five wire interface commonly found on many PC sound cards. It is
+now also popular in many portable devices. This DAI has a reset line and time
+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
+frame is 21uS long and is divided into 13 time slots.
+
+The AC97 specification can be found at http://intel.com/
+
+
+I2S
+===
+
+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmision, whilst the bit clock (BCLK) and
+left/right clock (LRC) synchronise the link. I2S is flexible in that either the
+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
+usually varies depending on the sample rate and the master system clock
+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
+ADC and DAC LRCLK's, this allows for similtanious capture and playback at
+different sample rates.
+
+I2S has several different operating modes:-
+
+ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
+ transition.
+
+ o Left Justified - MSB is transmitted on transition of LRC.
+
+ o Right Justified - MSB is transmitted sample size BCLK's before LRC
+ transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, that can support a more
+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
+to synchronise the link whilst the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+whilst sync runs at the sample rate. PCM also supports Time Division
+Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This
+is sometimes referred to as network mode).
+
+Common PCM operating modes:-
+
+ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
+
+
+ASoC DAI Configuration
+======================
+
+Every CODEC DAI and SoC DAI must have their capabilities defined in order to
+be configured together at runtime when the audio and clocking parameters are
+known. This is achieved by creating an array of struct snd_soc_hw_mode in the
+the CODEC and SoC interface drivers. Each element in the array describes a DAI
+mode and each mode is usually based upon the DAI system clock to sample rate
+ratio (FS).
+
+i.e. 48k sample rate @ 256 FS = sytem clock of 12.288 MHz
+ 48000 * 256 = 12288000
+
+The CPU and Codec DAI modes are then ANDed together at runtime to determine the
+rutime DAI configuration for both the Codec and CPU.
+
+When creating a new codec or SoC DAI it's probably best to start of with a few
+sample rates first and then test your interface.
+
+struct snd_soc_dai_mode is defined (in soc.h) as:-
+
+/* SoC DAI mode */
+struct snd_soc_hw_mode {
+ unsigned int fmt:16; /* SND_SOC_DAIFMT_* */
+ unsigned int tdm:16; /* SND_SOC_DAITDM_* */
+ unsigned int pcmfmt:6; /* SNDRV_PCM_FORMAT_* */
+ unsigned int pcmrate:16; /* SND_SOC_DAIRATE_* */
+ unsigned int pcmdir:2; /* SND_SOC_DAIDIR_* */
+ unsigned int flags:8; /* hw flags */
+ unsigned int fs:32; /* mclk to rate dividers */
+ unsigned int bfs:16; /* mclk to bclk dividers */
+ unsigned long priv; /* private mode data */
+};
+
+fmt:
+----
+This field defines the DAI mode hardware format (e.g. I2S settings) and
+supports the following settings:-
+
+ 1) hardware DAI formats
+
+#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */
+#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */
+#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */
+#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM */
+#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM */
+#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */
+
+ 2) hw DAI signal inversions
+
+#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */
+#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */
+#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */
+
+ 3) hw clock masters
+ This is wrt the codec, the inverse is true for the interface
+ i.e. if the codec is clk and frm master then the interface is
+ clk and frame slave.
+
+#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */
+#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */
+#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */
+
+At least one option from each section must be selected. Multiple selections are
+also supported e.g.
+
+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
+ SND_SOC_DAIFMT_IB_IF
+
+
+tdm:
+------
+This field defines the Time Division Multiplexing left and right word
+positions for the DAI mode if applicable. Set to SND_SOC_DAITDM_LRDW(0,0) for
+no TDM.
+
+
+pcmfmt:
+---------
+The hardware PCM format. This describes the PCM formats supported by the DAI
+mode e.g.
+
+ .hwpcmfmt = SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
+ SNDRV_PCM_FORMAT_S24_3LE
+
+pcmrate:
+----------
+The PCM sample rates supported by the DAI mode. e.g.
+
+ .hwpcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000
+
+
+pcmdir:
+---------
+The stream directions supported by this mode. e.g. playback and capture
+
+
+flags:
+--------
+The DAI hardware flags supported by the mode.
+
+SND_SOC_DAI_BFS_DIV
+This flag states that bit clock is generated by dividing MCLK in this mode, if
+this flag is absent the bitclock generated by mulitiplying sample rate.
+
+NOTE: Bitclock division and mulitiplication modes can be safely matched by the
+core logic.
+
+
+fs:
+-----
+The FS supported by this DAI mode FS is the ratio between the system clock and
+the sample rate. See above
+
+bfs:
+------
+BFS is the ratio of BCLK to MCLK or the ratio of BCLK to sample rate (this
+depends on the codec or CPU DAI).
+
+The BFS supported by the DAI mode. This can either be the ratio between the
+bitclock (BCLK) and the sample rate OR the ratio between the system clock and
+the sample rate. Depends on the SND_SOC_DAI_BFS_DIV flag above.
+
+priv:
+-----
+private codec mode data.
+
+
+
+Examples
+========
+
+Note that Codec DAI and CPU DAI examples are interchangeable in these examples
+as long as the bus master is reversed. i.e.
+
+ SND_SOC_DAIFMT_CBM_CFM would become SND_SOC_DAIFMT_CBS_CFS
+ and vice versa.
+
+This applies to all SND_SOC_DAIFMT_CB*_CF*.
+
+Example 1
+---------
+
+Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
+BCLK of either MCLK/2 or MCLK/4.
+
+ /* codec master */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 256, SND_SOC_FSBD(2) | SND_SOC_FSBD(4)},
+
+
+Example 2
+---------
+Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
+BCLK of either Rate * 32 or Rate * 64.
+
+ /* codec master */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
+ 256, SND_SOC_FSB(32) | SND_SOC_FSB(64)},
+
+
+Example 3
+---------
+Codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
+BCLK of either Rate * 32 or Rate * 64. Codec can also run in slave mode as long
+as BCLK is rate * 32 or rate * 64.
+
+ /* codec master */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
+ 256, SND_SOC_FSB(32) | SND_SOC_FSB(64)},
+
+ /* codec slave */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
+ SND_SOC_FS_ALL, SND_SOC_FSB(32) | SND_SOC_FSB(64)},
+
+
+Example 4
+---------
+Codec that only runs at 8k, 16k, 32k, 48k, 96k @ 128FS, 192FS & 256FS in master
+mode and can generate a BCLK of MCLK / (1,2,4,8,16). Codec can also run in slave
+mode as and does not care about FS or BCLK (as long as there is enough bandwidth).
+
+ #define CODEC_FSB \
+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
+
+ #define CODEC_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+ /* codec master @ 128, 192 & 256 FS */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 128, CODEC_FSB},
+
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 192, CODEC_FSB},
+
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 256, CODEC_FSB},
+
+ /* codec slave */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
+ SND_SOC_FS_ALL, SND_SOC_FSB_ALL},
+
+
+Example 5
+---------
+Codec that only runs at 8k, 44.1k, 48k @ different FS in master mode (for use
+with a fixed MCLK) and can generate a BCLK of MCLK / (1,2,4,8,16).
+Codec can also run in slave mode as and does not care about FS or BCLK (as long
+as there is enough bandwidth). Codec can support 16, 24 and 32 bit PCM sample
+sizes.
+
+ #define CODEC_FSB \
+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
+
+ #define CODEC_PCM_FORMATS \
+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
+ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | SNDRV_PCM_FORMAT_S32_LE)
+
+ /* codec master */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 1536, CODEC_FSB},
+
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_44100,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 272, CODEC_FSB},
+
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_48000,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
+ 256, CODEC_FSB},
+
+ /* codec slave */
+ {SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0),
+ SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
+ SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
+ SND_SOC_FS_ALL, SND_SOC_FSB_ALL},
+
+
+Example 6
+---------
+AC97 Codec that does not support VRA (i.e only runs at 48k).
+
+ #define AC97_DIR \
+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+
+ #define AC97_PCM_FORMATS \
+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S18_3LE | \
+ SNDRV_PCM_FORMAT_S20_3LE)
+
+ /* AC97 with no VRA */
+ {0, 0, AC97_PCM_FORMATS, SNDRV_PCM_RATE_48000},
+
+
+Example 7
+---------
+
+CPU DAI that supports 8k - 48k @ 256FS and BCLK = MCLK / 4 in master mode.
+Slave mode (CPU DAI is FRAME master) supports 8k - 96k at any FS as long as
+BCLK = 64 * rate. (Intel XScale I2S controller).
+
+ #define PXA_I2S_DAIFMT \
+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF)
+
+ #define PXA_I2S_DIR \
+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+ #define PXA_I2S_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+ /* pxa2xx I2S frame and clock master modes */
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_RATE_8000, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
+ SND_SOC_FSBD(4), 0x48},
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_RATE_11025, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
+ SND_SOC_FSBD(4), 0x34},
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_RATE_16000, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
+ SND_SOC_FSBD(4), 0x24},
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_RATE_22050, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
+ SND_SOC_FSBD(4), 0x1a},
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_RATE_44100, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
+ SND_SOC_FSBD(4), 0xd},
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_RATE_48000, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
+ SND_SOC_FSBD(4), 0xc},
+
+ /* pxa2xx I2S frame master and clock slave mode */
+ {PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
+ PXA_I2S_RATES, PXA_I2S_DIR, 0, SND_SOC_FS_ALL, SND_SOC_FSB(64)},
+
diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
new file mode 100644
index 0000000..88a16c9
--- /dev/null
+++ b/Documentation/sound/alsa/soc/clocking.txt
@@ -0,0 +1,309 @@
+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);
+ }
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
new file mode 100644
index 0000000..47b36cb
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,232 @@
+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;
+}
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
new file mode 100644
index 0000000..c11877f
--- /dev/null
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -0,0 +1,297 @@
+Dynamic Audio Power Management for Portable Devices
+===================================================
+
+1. Description
+==============
+
+Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices
+to use the minimum amount of power within the audio subsystem at all times. It
+is independent of other kernel PM and as such, can easily co-exist with the
+other PM systems.
+
+DAPM is also completely transparent to all user space applications as all power
+switching is done within the ASoC core. No code changes or recompiling are
+required for user space applications. DAPM makes power switching descisions based
+upon any audio stream (capture/playback) activity and audio mixer settings
+within the device.
+
+DAPM spans the whole machine. It covers power control within the entire audio
+subsystem, this includes internal codec power blocks and machine level power
+systems.
+
+There are 4 power domains within DAPM
+
+ 1. Codec domain - VREF, VMID (core codec and audio power)
+ Usually controlled at codec probe/remove and suspend/resume, although
+ can be set at stream time if power is not needed for sidetone, etc.
+
+ 2. Platform/Machine domain - physically connected inputs and outputs
+ Is platform/machine and user action specific, is configured by the
+ machine driver and responds to asynchronous events e.g when HP
+ are inserted
+
+ 3. Path domain - audio susbsystem signal paths
+ Automatically set when mixer and mux settings are changed by the user.
+ e.g. alsamixer, amixer.
+
+ 4. Stream domain - DAC's and ADC's.
+ Enabled and disabled when stream playback/capture is started and
+ stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching descisons are made automatically by consulting an audio
+routing map of the whole machine. This map is specific to each machine and
+consists of the interconnections between every audio component (including
+internal codec components). All audio components that effect power are called
+widgets hereafter.
+
+
+2. DAPM Widgets
+===============
+
+Audio DAPM widgets fall into a number of types:-
+
+ o Mixer - Mixes several analog signals into a single analog signal.
+ o Mux - An analog switch that outputs only 1 of it's inputs.
+ o PGA - A programmable gain amplifier or attenuation widget.
+ o ADC - Analog to Digital Converter
+ o DAC - Digital to Analog Converter
+ o Switch - An analog switch
+ o Input - A codec input pin
+ o Output - A codec output pin
+ o Headphone - Headphone (and optional Jack)
+ o Mic - Mic (and optional Jack)
+ o Line - Line Input/Output (and optional Jack)
+ o Speaker - Speaker
+ o Pre - Special PRE widget (exec before all others)
+ o Post - Special POST widget (exec after all others)
+
+(Widgets are defined in include/sound/soc-dapm.h)
+
+Widgets are usually added in the codec driver and the machine driver. There are
+convience macros defined in soc-dapm.h that can be used to quickly build a
+list of widgets of the codecs and machines DAPM widgets.
+
+Most widgets have a name, register, shift and invert. Some widgets have extra
+parameters for stream name and kcontrols.
+
+
+2.1 Stream Domain Widgets
+-------------------------
+
+Stream Widgets relate to the stream power domain and only consist of ADC's
+(analog to digital converters) and DAC's (digital to analog converters).
+
+Stream widgets have the following format:-
+
+SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+
+NOTE: the stream name must match the corresponding stream name in your codecs
+snd_soc_codec_dai.
+
+e.g. stream widgets for HiFi playback and capture
+
+SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
+SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
+
+
+2.2 Path Domain Widgets
+-----------------------
+
+Path domain widgets have a ability to control or effect the audio signal or
+audio paths within the audio subsystem. They have the following form:-
+
+SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
+
+Any widget kcontrols can be set using the controls and num_controls members.
+
+e.g. Mixer widget (the kcontrols are declared first)
+
+/* Output Mixer */
+static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
+ ARRAY_SIZE(wm8731_output_mixer_controls)),
+
+
+2.3 Platform/Machine domain Widgets
+-----------------------------------
+
+Machine widgets are different from codec widgets in that they don't have a
+codec register bit associated with them. A machine widget is assigned to each
+machine audio component (non codec) that can be independently powered. e.g.
+
+ o Speaker Amp
+ o Microphone Bias
+ o Jack connectors
+
+A machine widget can have an optional call back.
+
+e.g. Jack connector widget for an external Mic that enables Mic Bias
+when the Mic is inserted:-
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
+{
+ if(SND_SOC_DAPM_EVENT_ON(event))
+ set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
+ else
+ reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
+
+ return 0;
+}
+
+SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+
+
+2.4 Codec Domain
+----------------
+
+The Codec power domain has no widgets and is handled by the codecs DAPM event
+handler. This handler is called when the codec powerstate is changed wrt to any
+stream event or by kernel PM events.
+
+
+2.5 Virtual Widgets
+-------------------
+
+Sometimes widgets exist in the codec or machine audio map that don't have any
+corresponding register bit for power control. In this case it's necessary to
+create a virtual widget - a widget with no control bits e.g.
+
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
+
+This can be used to merge to signal paths together in software.
+
+After all the widgets have been defined, they can then be added to the DAPM
+subsystem individually with a call to snd_soc_dapm_new_control().
+
+
+3. Codec Widget Interconnections
+================================
+
+Widgets are connected to each other within the codec and machine by audio
+paths (called interconnections). Each interconnection must be defined in order
+to create a map of all audio paths between widgets.
+This is easiest with a diagram of the codec (and schematic of the machine audio
+system), as it requires joining widgets together via their audio signal paths.
+
+i.e. from the WM8731 codec's output mixer (wm8731.c)
+
+The WM8731 output mixer has 3 inputs (sources)
+
+ 1. Line Bypass Input
+ 2. DAC (HiFi playback)
+ 3. Mic Sidetone Input
+
+Each input in this example has a kcontrol associated with it (defined in example
+above) and is connected to the output mixer via it's kcontrol name. We can now
+connect the destination widget (wrt audio signal) with it's source widgets.
+
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+So we have :-
+
+ Destination Widget <=== Path Name <=== Source Widget
+
+Or:-
+
+ Sink, Path, Source
+
+Or :-
+
+ "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".
+
+When there is no path name connecting widgets (e.g. a direct connection) we
+pass NULL for the path name.
+
+Interconnections are created with a call to:-
+
+snd_soc_dapm_connect_input(codec, sink, path, source);
+
+Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
+interconnections have been registered with the core. This causes the core to
+scan the codec and machine so that the internal DAPM state matches the
+physical state of the machine.
+
+
+3.1 Machine Widget Interconnections
+-----------------------------------
+Machine widget interconnections are created in the same way as codec ones and
+directly connect the codec pins to machine level widgets.
+
+e.g. connects the speaker out codec pins to the internal speaker.
+
+ /* ext speaker connected to codec pins LOUT2, ROUT2 */
+ {"Ext Spk", NULL , "ROUT2"},
+ {"Ext Spk", NULL , "LOUT2"},
+
+This allows the DAPM to power on and off pins that are connected (and in use)
+and pins that are NC respectively.
+
+
+4 Endpoint Widgets
+===================
+An endpoint is a start or end point (widget) of an audio signal within the
+machine and includes the codec. e.g.
+
+ o Headphone Jack
+ o Internal Speaker
+ o Internal Mic
+ o Mic Jack
+ o Codec Pins
+
+When a codec pin is NC it can be marked as not used with a call to
+
+snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
+
+The last argument is 0 for inactive and 1 for active. This way the pin and its
+input widget will never be powered up and consume power.
+
+This also applies to machine widgets. e.g. if a headphone is connected to a
+jack then the jack can be marked active. If the headphone is removed, then
+the headphone jack can be marked inactive.
+
+
+5 DAPM Widget Events
+====================
+
+Some widgets can register their interest with the DAPM core in PM events.
+e.g. A Speaker with an amplifier registers a widget so the amplifier can be
+powered only when the spk is in use.
+
+/* turn speaker amplifier on/off depending on use */
+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+ else
+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+
+ return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
+ SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
+
+Please see soc-dapm.h for all other widgets that support events.
+
+
+5.1 Event types
+---------------
+
+The following event types are supported by event widgets.
+
+/* dapm event types */
+#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
+#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
+#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
+#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
+#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
+#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
new file mode 100644
index 0000000..3014795
--- /dev/null
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -0,0 +1,114 @@
+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
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
new file mode 100644
index 0000000..753c5cc
--- /dev/null
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -0,0 +1,83 @@
+ALSA SoC Layer
+==============
+
+The overall project goal of the ALSA System on Chip (ASoC) layer is to provide
+better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00,
+iMX, etc) and portable audio codecs. Currently there is some support in the
+kernel for SoC audio, however it has some limitations:-
+
+ * Currently, codec drivers are often tightly coupled to the underlying SoC
+ cpu. This is not ideal and leads to code duplication i.e. Linux now has 4
+ different wm8731 drivers for 4 different SoC platforms.
+
+ * There is no standard method to signal user initiated audio events.
+ e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion
+ event. These are quite common events on portable devices and ofter require
+ machine specific code to re route audio, enable amps etc after such an event.
+
+ * Current drivers tend to power up the entire codec when playing
+ (or recording) audio. This is fine for a PC, but tends to waste a lot of
+ power on portable devices. There is also no support for saving power via
+ changing codec oversampling rates, bias currents, etc.
+
+
+ASoC Design
+===========
+
+The ASoC layer is designed to address these issues and provide the following
+features :-
+
+ * Codec independence. Allows reuse of codec drivers on other platforms
+ and machines.
+
+ * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface
+ and codec registers it's audio interface capabilities with the core and are
+ subsequently matched and configured when the application hw params are known.
+
+ * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+ it's minimum power state at all times. This includes powering up/down
+ internal power blocks depending on the internal codec audio routing and any
+ active streams.
+
+ * Pop and click reduction. Pops and clicks can be reduced by powering the
+ codec up/down in the correct sequence (including using digital mute). ASoC
+ signals the codec when to change power states.
+
+ * Machine specific controls: Allow machines to add controls to the sound card
+ e.g. volume control for speaker amp.
+
+To achieve all this, ASoC basically splits an embedded audio system into 3
+components :-
+
+ * Codec driver: The codec driver is platform independent and contains audio
+ controls, audio interface capabilities, codec dapm definition and codec IO
+ functions.
+
+ * Platform driver: The platform driver contains the audio dma engine and audio
+ interface drivers (e.g. I2S, AC97, PCM) for that platform.
+
+ * Machine driver: The machine driver handles any machine specific controls and
+ audio events. i.e. turing on an amp at start of playback.
+
+
+Documentation
+=============
+
+The documentation is spilt into the following sections:-
+
+overview.txt: This file.
+
+codec.txt: Codec driver internals.
+
+DAI.txt: Description of Digital Audio Interface standards and how to configure
+a DAI within your codec and CPU DAI drivers.
+
+dapm.txt: Dynamic Audio Power Management
+
+platform.txt: Platform audio DMA and DAI.
+
+machine.txt: Machine driver internals.
+
+pop_clicks.txt: How to minimise audio artifacts.
+
+clocking.txt: ASoC clocking for best power performance.
\ No newline at end of file
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
new file mode 100644
index 0000000..c88df26
--- /dev/null
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -0,0 +1,58 @@
+ASoC Platform Driver
+====================
+
+An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
+and control. The platform drivers only target the SoC CPU and must have no board
+specific code.
+
+Audio DMA
+=========
+
+The platform DMA driver optionally supports the following alsa operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+ int (*startup)(snd_pcm_substream_t *);
+ void (*shutdown)(snd_pcm_substream_t *);
+ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *);
+ int (*hw_free)(snd_pcm_substream_t *);
+ int (*prepare)(snd_pcm_substream_t *);
+ int (*trigger)(snd_pcm_substream_t *, int);
+};
+
+The platform driver exports it's DMA functionailty via struct snd_soc_platform:-
+
+struct snd_soc_platform {
+ char *name;
+
+ int (*probe)(struct platform_device *pdev);
+ int (*remove)(struct platform_device *pdev);
+ int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+ int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+
+ /* pcm creation and destruction */
+ int (*pcm_new)(snd_card_t *, struct snd_soc_codec_dai *, snd_pcm_t *);
+ void (*pcm_free)(snd_pcm_t *);
+
+ /* platform stream ops */
+ snd_pcm_ops_t *pcm_ops;
+};
+
+Please refer to the alsa driver documentation for details of audio DMA.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
+
+An example DMA driver is soc/pxa/pxa2xx-pcm.c
+
+
+SoC DAI Drivers
+===============
+
+Each SoC DAI driver must provide the following features:-
+
+ 1) Digital audio interface (DAI) description
+ 2) Digital audio interface configuration
+ 3) PCM's description
+ 4) Sysclk configuration
+ 5) Suspend and resume (optional)
+
+Please see codec.txt for a description of items 1 - 4.
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
new file mode 100644
index 0000000..f4f8de5
--- /dev/null
+++ b/Documentation/sound/alsa/soc/pops_clicks.txt
@@ -0,0 +1,52 @@
+Audio Pops and Clicks
+=====================
+
+Pops and clicks are unwanted audio artifacts caused by the powering up and down
+of components within the audio subsystem. This is 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.
diff --git a/MAINTAINERS b/MAINTAINERS
index fe35f3a..010658a 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3037,6 +3037,12 @@
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