[ALSA] ASoC documentation updates
This patch updates the documentation for ASoC to reflect the recent changes in API between 0.12.x and 0.13.x Changes:- o Removed all reference to old API's. o Removed references and examples of automatic DAI config and matching. o Fixed 80 char line length on some files. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
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5 changed files with 48 additions and 837 deletions
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@ -54,493 +54,3 @@ 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_dai_mode {
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u16 fmt; /* SND_SOC_DAIFMT_* */
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u16 tdm; /* SND_SOC_HWTDM_* */
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u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */
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u16 pcmrate; /* SND_SOC_HWRATE_* */
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u16 pcmdir:2; /* SND_SOC_HWDIR_* */
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u16 flags:8; /* hw flags */
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u16 fs; /* mclk to rate divider */
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u64 bfs; /* 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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.pcmfmt = 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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.pcmrate = 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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/* use bfs mclk divider mode (BCLK = MCLK / x) */
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#define SND_SOC_DAI_BFS_DIV 0x1
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/* use bfs rate mulitplier (BCLK = RATE * x)*/
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#define SND_SOC_DAI_BFS_RATE 0x2
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/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */
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#define SND_SOC_DAI_BFS_RCW 0x4
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/* capture and playback can use different clocks */
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#define SND_SOC_DAI_ASYNC 0x8
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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 flags 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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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 256,
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.bfs = SND_SOC_FSBD(2) | SND_SOC_FSBD(4),
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}
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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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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RATE,
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.fs = 256,
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.bfs = 32,
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RATE,
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.fs = 256,
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.bfs = 64,
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},
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Example 3
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---------
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Codec that runs at 8k & 48k @ 256FS in master mode, can generate a BCLK that
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is a multiple of Rate * channels * word size. (RCW) i.e.
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BCLK = 8000 * 2 * 16 (8k, stereo, 16bit)
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= 256kHz
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This codecs supports a RCW multiple of 1,2
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RCW,
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.fs = 256,
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.bfs = SND_SOC_FSBW(1) | SND_SOC_FSBW(2),
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}
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Example 4
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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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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RATE,
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.fs = 256,
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.bfs = 32,
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RATE,
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.fs = 256,
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.bfs = 64,
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},
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/* codec slave */
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RATE,
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.fs = SND_SOC_FS_ALL,
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.bfs = 32,
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_RATE,
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.fs = SND_SOC_FS_ALL,
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.bfs = 64,
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},
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Example 5
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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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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = CODEC_RATES,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 128,
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.bfs = CODEC_FSB,
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = CODEC_RATES,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 192,
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.bfs = CODEC_FSB
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = CODEC_RATES,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 256,
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.bfs = CODEC_FSB,
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},
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/* codec slave */
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = CODEC_RATES,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.fs = SND_SOC_FS_ALL,
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.bfs = SND_SOC_FSB_ALL,
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},
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Example 6
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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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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_8000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 1536,
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.bfs = CODEC_FSB,
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_44100,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 272,
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.bfs = CODEC_FSB,
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},
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = SNDRV_PCM_RATE_48000,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.flags = SND_SOC_DAI_BFS_DIV,
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.fs = 256,
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.bfs = CODEC_FSB,
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},
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/* codec slave */
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{
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.fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
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.pcmfmt = SNDRV_PCM_FORMAT_S16_LE,
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.pcmrate = CODEC_RATES,
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.pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE,
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.fs = SND_SOC_FS_ALL,
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.bfs = SND_SOC_FSB_ALL,
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},
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Example 7
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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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{
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.pcmfmt = AC97_PCM_FORMATS,
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.pcmrate = SNDRV_PCM_RATE_48000,
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}
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Example 8
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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
|
||||
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)
|
||||
|
||||
/* priv is divider */
|
||||
static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = {
|
||||
/* pxa2xx I2S frame and clock master modes */
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = SNDRV_PCM_RATE_8000,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.flags = SND_SOC_DAI_BFS_DIV,
|
||||
.fs = 256,
|
||||
.bfs = SND_SOC_FSBD(4),
|
||||
.priv = 0x48,
|
||||
},
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = SNDRV_PCM_RATE_11025,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.flags = SND_SOC_DAI_BFS_DIV,
|
||||
.fs = 256,
|
||||
.bfs = SND_SOC_FSBD(4),
|
||||
.priv = 0x34,
|
||||
},
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = SNDRV_PCM_RATE_16000,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.flags = SND_SOC_DAI_BFS_DIV,
|
||||
.fs = 256,
|
||||
.bfs = SND_SOC_FSBD(4),
|
||||
.priv = 0x24,
|
||||
},
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = SNDRV_PCM_RATE_22050,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.flags = SND_SOC_DAI_BFS_DIV,
|
||||
.fs = 256,
|
||||
.bfs = SND_SOC_FSBD(4),
|
||||
.priv = 0x1a,
|
||||
},
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = SNDRV_PCM_RATE_44100,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.flags = SND_SOC_DAI_BFS_DIV,
|
||||
.fs = 256,
|
||||
.bfs = SND_SOC_FSBD(4),
|
||||
.priv = 0xd,
|
||||
},
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = SNDRV_PCM_RATE_48000,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.flags = SND_SOC_DAI_BFS_DIV,
|
||||
.fs = 256,
|
||||
.bfs = SND_SOC_FSBD(4),
|
||||
.priv = 0xc,
|
||||
},
|
||||
|
||||
/* pxa2xx I2S frame master and clock slave mode */
|
||||
{
|
||||
.fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS,
|
||||
.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
|
||||
.pcmrate = PXA_I2S_RATES,
|
||||
.pcmdir = PXA_I2S_DIR,
|
||||
.fs = SND_SOC_FS_ALL,
|
||||
.flags = SND_SOC_DAI_BFS_RATE,
|
||||
.bfs = 64,
|
||||
.priv = 0x48,
|
||||
},
|
||||
};
|
||||
|
|
|
@ -40,275 +40,12 @@ BCLK = LRC * x
|
|||
|
||||
BCLK = LRC * Channels * Word Size
|
||||
|
||||
This relationship depends on the codec or SoC CPU in particular. ASoC can quite
|
||||
easily match BCLK generated by division (SND_SOC_DAI_BFS_DIV) with BCLK by
|
||||
multiplication (SND_SOC_DAI_BFS_RATE) or BCLK generated by
|
||||
Rate * Channels * Word size (RCW or SND_SOC_DAI_BFS_RCW).
|
||||
This relationship depends on the codec or SoC CPU in particular. In general
|
||||
it's best to configure BCLK to the lowest possible speed (depending on your
|
||||
rate, number of channels and wordsize) to save on power.
|
||||
|
||||
It's also desireable to use the codec (if possible) to drive (or master) the
|
||||
audio clocks as it's usually gives more accurate sample rates than the CPU.
|
||||
|
||||
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);
|
||||
}
|
||||
|
|
|
@ -6,21 +6,18 @@ 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:-
|
||||
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
|
||||
1) Codec DAI and PCM configuration
|
||||
2) Codec control IO - using I2C, 3 Wire(SPI) or both API's
|
||||
3) Mixers and audio controls
|
||||
4) Codec audio operations
|
||||
|
||||
Optionally, codec drivers can also provide:-
|
||||
|
||||
8) DAPM description.
|
||||
9) DAPM event handler.
|
||||
10) DAC Digital mute control.
|
||||
5) DAPM description.
|
||||
6) DAPM event handler.
|
||||
7) DAC Digital mute control.
|
||||
|
||||
It's probably best to use this guide in conjuction with the existing codec
|
||||
driver code in sound/soc/codecs/
|
||||
|
@ -28,58 +25,47 @@ 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.
|
||||
1 - Codec DAI and PCM configuration
|
||||
-----------------------------------
|
||||
Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and
|
||||
PCM's capablities and operations. This struct is exported so that it can be
|
||||
registered with the core by your machine driver.
|
||||
|
||||
e.g.
|
||||
|
||||
static struct snd_soc_pcm_codec wm8731_pcm_client = {
|
||||
struct snd_soc_codec_dai wm8731_dai = {
|
||||
.name = "WM8731",
|
||||
/* playback capabilities */
|
||||
.playback = {
|
||||
.stream_name = "Playback",
|
||||
.channels_min = 1,
|
||||
.channels_max = 2,
|
||||
},
|
||||
.rates = WM8731_RATES,
|
||||
.formats = WM8731_FORMATS,},
|
||||
/* capture capabilities */
|
||||
.capture = {
|
||||
.stream_name = "Capture",
|
||||
.channels_min = 1,
|
||||
.channels_max = 2,
|
||||
},
|
||||
.config_sysclk = wm8731_config_sysclk,
|
||||
.rates = WM8731_RATES,
|
||||
.formats = WM8731_FORMATS,},
|
||||
/* pcm operations - see section 4 below */
|
||||
.ops = {
|
||||
.prepare = wm8731_pcm_prepare,
|
||||
.hw_params = wm8731_hw_params,
|
||||
.shutdown = wm8731_shutdown,
|
||||
},
|
||||
.caps = {
|
||||
.num_modes = ARRAY_SIZE(wm8731_hwfmt),
|
||||
.modes = &wm8731_hwfmt[0],
|
||||
},
|
||||
/* DAI operations - see DAI.txt */
|
||||
.dai_ops = {
|
||||
.digital_mute = wm8731_mute,
|
||||
.set_sysclk = wm8731_set_dai_sysclk,
|
||||
.set_fmt = wm8731_set_dai_fmt,
|
||||
}
|
||||
};
|
||||
EXPORT_SYMBOL_GPL(wm8731_dai);
|
||||
|
||||
|
||||
4 - Codec control IO
|
||||
2 - 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
|
||||
|
@ -104,7 +90,7 @@ read/write:-
|
|||
hw_read_t hw_read;
|
||||
|
||||
|
||||
5 - Mixers and audio controls
|
||||
3 - Mixers and audio controls
|
||||
-----------------------------
|
||||
All the codec mixers and audio controls can be defined using the convenience
|
||||
macros defined in soc.h.
|
||||
|
@ -143,28 +129,7 @@ Defines an single enumerated control as follows:-
|
|||
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
|
||||
4 - Codec Audio Operations
|
||||
--------------------------
|
||||
The codec driver also supports the following alsa operations:-
|
||||
|
||||
|
@ -181,7 +146,7 @@ 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.
|
||||
5 - DAPM description.
|
||||
---------------------
|
||||
The Dynamic Audio Power Management description describes the codec's power
|
||||
components, their relationships and registers to the ASoC core. Please read
|
||||
|
@ -190,7 +155,7 @@ dapm.txt for details of building the description.
|
|||
Please also see the examples in other codec drivers.
|
||||
|
||||
|
||||
9 - DAPM event handler
|
||||
6 - 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
|
||||
|
@ -210,7 +175,7 @@ Power states:-
|
|||
SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
|
||||
|
||||
|
||||
10 - Codec DAC digital mute control.
|
||||
7 - 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.
|
||||
|
|
|
@ -64,7 +64,7 @@ static struct snd_soc_dai_link corgi_dai = {
|
|||
.cpu_dai = &pxa_i2s_dai,
|
||||
.codec_dai = &wm8731_dai,
|
||||
.init = corgi_wm8731_init,
|
||||
.config_sysclk = corgi_config_sysclk,
|
||||
.ops = &corgi_ops,
|
||||
};
|
||||
|
||||
struct snd_soc_machine then sets up the machine with it's DAI's. e.g.
|
||||
|
@ -74,7 +74,6 @@ static struct snd_soc_machine snd_soc_machine_corgi = {
|
|||
.name = "Corgi",
|
||||
.dai_link = &corgi_dai,
|
||||
.num_links = 1,
|
||||
.ops = &corgi_ops,
|
||||
};
|
||||
|
||||
|
||||
|
|
|
@ -2,14 +2,14 @@ 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
|
||||
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.
|
||||
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
|
||||
|
|
Loading…
Reference in a new issue