b5d5c87986
There are examples in documentation for codec to codec connection.
However they show method before recent series of patches which renamed
the fields. Update documentation accordingly.
Fixes: 7ddc7f91be
("ASoC: soc.h: clarify Codec2Codec params")
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230928134706.662947-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
116 lines
3.8 KiB
ReStructuredText
116 lines
3.8 KiB
ReStructuredText
==============================================
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Creating codec to codec dai link for ALSA dapm
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==============================================
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Mostly the flow of audio is always from CPU to codec so your system
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will look as below:
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::
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--------- ---------
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| | dai | |
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CPU -------> codec
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--------- ---------
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In case your system looks as below:
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::
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---------
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codec-2
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---------
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dai-2
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---------- ---------
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| | dai-1 | |
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CPU -------> codec-1
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---------- ---------
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dai-3
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---------
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codec-3
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---------
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Suppose codec-2 is a bluetooth chip and codec-3 is connected to
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a speaker and you have a below scenario:
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codec-2 will receive the audio data and the user wants to play that
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audio through codec-3 without involving the CPU.This
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aforementioned case is the ideal case when codec to codec
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connection should be used.
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Your dai_link should appear as below in your machine
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file:
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::
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/*
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* this pcm stream only supports 24 bit, 2 channel and
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* 48k sampling rate.
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*/
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static const struct snd_soc_pcm_stream dsp_codec_params = {
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.formats = SNDRV_PCM_FMTBIT_S24_LE,
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.rate_min = 48000,
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.rate_max = 48000,
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.channels_min = 2,
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.channels_max = 2,
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};
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{
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.name = "CPU-DSP",
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.stream_name = "CPU-DSP",
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.cpu_dai_name = "samsung-i2s.0",
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.codec_name = "codec-2,
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.codec_dai_name = "codec-2-dai_name",
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.platform_name = "samsung-i2s.0",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
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| SND_SOC_DAIFMT_CBM_CFM,
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.ignore_suspend = 1,
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.c2c_params = &dsp_codec_params,
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.num_c2c_params = 1,
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},
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{
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.name = "DSP-CODEC",
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.stream_name = "DSP-CODEC",
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.cpu_dai_name = "wm0010-sdi2",
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.codec_name = "codec-3,
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.codec_dai_name = "codec-3-dai_name",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
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| SND_SOC_DAIFMT_CBM_CFM,
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.ignore_suspend = 1,
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.c2c_params = &dsp_codec_params,
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.num_c2c_params = 1,
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},
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Above code snippet is motivated from sound/soc/samsung/speyside.c.
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Note the "c2c_params" callback which lets the dapm know that this
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dai_link is a codec to codec connection.
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In dapm core a route is created between cpu_dai playback widget
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and codec_dai capture widget for playback path and vice-versa is
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true for capture path. In order for this aforementioned route to get
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triggered, DAPM needs to find a valid endpoint which could be either
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a sink or source widget corresponding to playback and capture path
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respectively.
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In order to trigger this dai_link widget, a thin codec driver for
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the speaker amp can be created as demonstrated in wm8727.c file, it
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sets appropriate constraints for the device even if it needs no control.
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Make sure to name your corresponding cpu and codec playback and capture
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dai names ending with "Playback" and "Capture" respectively as dapm core
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will link and power those dais based on the name.
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A dai_link in a "simple-audio-card" will automatically be detected as
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codec to codec when all DAIs on the link belong to codec components.
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The dai_link will be initialized with the subset of stream parameters
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(channels, format, sample rate) supported by all DAIs on the link. Since
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there is no way to provide these parameters in the device tree, this is
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mostly useful for communication with simple fixed-function codecs, such
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as a Bluetooth controller or cellular modem.
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