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[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>
This commit is contained in:
Liam Girdwood
Jaroslav Kysela
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Documentation/sound/alsa/soc/DAI.txt
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380
Documentation/sound/alsa/soc/DAI.txt
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ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
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SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM.
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AC97
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====
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AC97 is a five wire interface commonly found on many PC sound cards. It is
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now also popular in many portable devices. This DAI has a reset line and time
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multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
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The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
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frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
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frame is 21uS long and is divided into 13 time slots.
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The AC97 specification can be found at http://intel.com/
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I2S
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===
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I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
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Rx lines are used for audio transmision, whilst the bit clock (BCLK) and
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left/right clock (LRC) synchronise the link. I2S is flexible in that either the
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controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
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usually varies depending on the sample rate and the master system clock
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(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
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ADC and DAC LRCLK's, this allows for similtanious capture and playback at
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different sample rates.
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I2S has several different operating modes:-
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o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
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transition.
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o Left Justified - MSB is transmitted on transition of LRC.
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o Right Justified - MSB is transmitted sample size BCLK's before LRC
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transition.
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PCM
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===
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PCM is another 4 wire interface, very similar to I2S, that can support a more
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flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
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to synchronise the link whilst the Tx and Rx lines are used to transmit and
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receive the audio data. Bit clock usually varies depending on sample rate
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whilst sync runs at the sample rate. PCM also supports Time Division
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Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This
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is sometimes referred to as network mode).
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Common PCM operating modes:-
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o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
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o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
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ASoC DAI Configuration
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======================
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Every CODEC DAI and SoC DAI must have their capabilities defined in order to
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be configured together at runtime when the audio and clocking parameters are
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known. This is achieved by creating an array of struct snd_soc_hw_mode in the
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the CODEC and SoC interface drivers. Each element in the array describes a DAI
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mode and each mode is usually based upon the DAI system clock to sample rate
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ratio (FS).
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i.e. 48k sample rate @ 256 FS = sytem clock of 12.288 MHz
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48000 * 256 = 12288000
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The CPU and Codec DAI modes are then ANDed together at runtime to determine the
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rutime DAI configuration for both the Codec and CPU.
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When creating a new codec or SoC DAI it's probably best to start of with a few
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sample rates first and then test your interface.
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struct snd_soc_dai_mode is defined (in soc.h) as:-
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/* SoC DAI mode */
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struct snd_soc_hw_mode {
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unsigned int fmt:16; /* SND_SOC_DAIFMT_* */
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unsigned int tdm:16; /* SND_SOC_DAITDM_* */
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unsigned int pcmfmt:6; /* SNDRV_PCM_FORMAT_* */
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unsigned int pcmrate:16; /* SND_SOC_DAIRATE_* */
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unsigned int pcmdir:2; /* SND_SOC_DAIDIR_* */
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unsigned int flags:8; /* hw flags */
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unsigned int fs:32; /* mclk to rate dividers */
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unsigned int bfs:16; /* mclk to bclk dividers */
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unsigned long priv; /* private mode data */
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};
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fmt:
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----
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This field defines the DAI mode hardware format (e.g. I2S settings) and
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supports the following settings:-
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1) hardware DAI formats
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#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */
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#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */
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#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */
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#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM */
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#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM */
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#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */
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2) hw DAI signal inversions
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#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */
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#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */
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#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */
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#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */
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3) hw clock masters
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This is wrt the codec, the inverse is true for the interface
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i.e. if the codec is clk and frm master then the interface is
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clk and frame slave.
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#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */
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#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */
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#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */
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#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */
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At least one option from each section must be selected. Multiple selections are
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also supported e.g.
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
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SND_SOC_DAIFMT_IB_IF
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tdm:
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------
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This field defines the Time Division Multiplexing left and right word
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positions for the DAI mode if applicable. Set to SND_SOC_DAITDM_LRDW(0,0) for
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no TDM.
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pcmfmt:
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---------
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The hardware PCM format. This describes the PCM formats supported by the DAI
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mode e.g.
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.hwpcmfmt = SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
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SNDRV_PCM_FORMAT_S24_3LE
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pcmrate:
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----------
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The PCM sample rates supported by the DAI mode. e.g.
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.hwpcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
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SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
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SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000
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pcmdir:
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---------
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The stream directions supported by this mode. e.g. playback and capture
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flags:
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--------
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The DAI hardware flags supported by the mode.
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SND_SOC_DAI_BFS_DIV
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This flag states that bit clock is generated by dividing MCLK in this mode, if
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this flag is absent the bitclock generated by mulitiplying sample rate.
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NOTE: Bitclock division and mulitiplication modes can be safely matched by the
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core logic.
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fs:
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-----
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The FS supported by this DAI mode FS is the ratio between the system clock and
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the sample rate. See above
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bfs:
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------
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BFS is the ratio of BCLK to MCLK or the ratio of BCLK to sample rate (this
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depends on the codec or CPU DAI).
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The BFS supported by the DAI mode. This can either be the ratio between the
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bitclock (BCLK) and the sample rate OR the ratio between the system clock and
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the sample rate. Depends on the SND_SOC_DAI_BFS_DIV flag above.
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priv:
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-----
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private codec mode data.
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Examples
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========
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Note that Codec DAI and CPU DAI examples are interchangeable in these examples
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as long as the bus master is reversed. i.e.
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SND_SOC_DAIFMT_CBM_CFM would become SND_SOC_DAIFMT_CBS_CFS
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and vice versa.
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This applies to all SND_SOC_DAIFMT_CB*_CF*.
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Example 1
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---------
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Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
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BCLK of either MCLK/2 or MCLK/4.
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/* codec master */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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256, SND_SOC_FSBD(2) | SND_SOC_FSBD(4)},
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Example 2
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---------
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Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
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BCLK of either Rate * 32 or Rate * 64.
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/* codec master */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
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256, SND_SOC_FSB(32) | SND_SOC_FSB(64)},
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Example 3
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---------
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Codec that only runs at 8k & 48k @ 256FS in master mode, can generate a
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BCLK of either Rate * 32 or Rate * 64. Codec can also run in slave mode as long
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as BCLK is rate * 32 or rate * 64.
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/* codec master */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
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256, SND_SOC_FSB(32) | SND_SOC_FSB(64)},
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/* codec slave */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
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SND_SOC_FS_ALL, SND_SOC_FSB(32) | SND_SOC_FSB(64)},
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Example 4
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---------
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Codec that only runs at 8k, 16k, 32k, 48k, 96k @ 128FS, 192FS & 256FS in master
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mode and can generate a BCLK of MCLK / (1,2,4,8,16). Codec can also run in slave
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mode as and does not care about FS or BCLK (as long as there is enough bandwidth).
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#define CODEC_FSB \
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(SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
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SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
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#define CODEC_RATES \
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(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
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SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
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/* codec master @ 128, 192 & 256 FS */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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128, CODEC_FSB},
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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192, CODEC_FSB},
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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256, CODEC_FSB},
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/* codec slave */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
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SND_SOC_FS_ALL, SND_SOC_FSB_ALL},
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Example 5
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---------
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Codec that only runs at 8k, 44.1k, 48k @ different FS in master mode (for use
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with a fixed MCLK) and can generate a BCLK of MCLK / (1,2,4,8,16).
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Codec can also run in slave mode as and does not care about FS or BCLK (as long
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as there is enough bandwidth). Codec can support 16, 24 and 32 bit PCM sample
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sizes.
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#define CODEC_FSB \
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(SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \
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SND_SOC_FSBD(8) | SND_SOC_FSBD(16))
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#define CODEC_PCM_FORMATS \
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(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
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SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | SNDRV_PCM_FORMAT_S32_LE)
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/* codec master */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_8000,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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1536, CODEC_FSB},
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_44100,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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272, CODEC_FSB},
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_RATE_48000,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, SND_SOC_DAI_BFS_DIV,
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256, CODEC_FSB},
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/* codec slave */
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{SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0),
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SNDRV_PCM_FORMAT_S16_LE, CODEC_RATES,
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SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, 0,
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SND_SOC_FS_ALL, SND_SOC_FSB_ALL},
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Example 6
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---------
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AC97 Codec that does not support VRA (i.e only runs at 48k).
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#define AC97_DIR \
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(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
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#define AC97_PCM_FORMATS \
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(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S18_3LE | \
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SNDRV_PCM_FORMAT_S20_3LE)
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/* AC97 with no VRA */
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{0, 0, AC97_PCM_FORMATS, SNDRV_PCM_RATE_48000},
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Example 7
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---------
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CPU DAI that supports 8k - 48k @ 256FS and BCLK = MCLK / 4 in master mode.
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Slave mode (CPU DAI is FRAME master) supports 8k - 96k at any FS as long as
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BCLK = 64 * rate. (Intel XScale I2S controller).
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#define PXA_I2S_DAIFMT \
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(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF)
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#define PXA_I2S_DIR \
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(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
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#define PXA_I2S_RATES \
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(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
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SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
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SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
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/* pxa2xx I2S frame and clock master modes */
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{PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
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SNDRV_PCM_RATE_8000, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
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SND_SOC_FSBD(4), 0x48},
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{PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE,
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SNDRV_PCM_RATE_11025, PXA_I2S_DIR, SND_SOC_DAI_BFS_DIV, 256,
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SND_SOC_FSBD(4), 0x34},
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{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)},
|
||||
|
||||
309
Documentation/sound/alsa/soc/clocking.txt
Normal file
309
Documentation/sound/alsa/soc/clocking.txt
Normal file
@ -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);
|
||||
}
|
||||
232
Documentation/sound/alsa/soc/codec.txt
Normal file
232
Documentation/sound/alsa/soc/codec.txt
Normal file
@ -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;
|
||||
}
|
||||
297
Documentation/sound/alsa/soc/dapm.txt
Normal file
297
Documentation/sound/alsa/soc/dapm.txt
Normal file
@ -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 */
|
||||
114
Documentation/sound/alsa/soc/machine.txt
Normal file
114
Documentation/sound/alsa/soc/machine.txt
Normal file
@ -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.
|
||||
83
Documentation/sound/alsa/soc/overview.txt
Normal file
83
Documentation/sound/alsa/soc/overview.txt
Normal file
@ -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.
|
||||
58
Documentation/sound/alsa/soc/platform.txt
Normal file
58
Documentation/sound/alsa/soc/platform.txt
Normal file
@ -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.
|
||||
52
Documentation/sound/alsa/soc/pops_clicks.txt
Normal file
52
Documentation/sound/alsa/soc/pops_clicks.txt
Normal file
@ -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.
|
||||
@ -3037,6 +3037,12 @@ M: perex@suse.cz
|
||||
L: alsa-devel@alsa-project.org
|
||||
S: Maintained
|
||||
|
||||
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
|
||||
P: Liam Girdwood
|
||||
M: liam.girdwood@wolfsonmicro.com
|
||||
L: alsa-devel@alsa-project.org
|
||||
S: Supported
|
||||
|
||||
SPI SUBSYSTEM
|
||||
P: David Brownell
|
||||
M: dbrownell@users.sourceforge.net
|
||||
|
||||
Loading…
Reference in New Issue
Block a user