forked from Minki/linux
95cfc0a0aa
Following the example in cb2cf0de11 ("ASoC: soc-core: care Codec <->
Codec case by non_legacy_dai_naming"), determine if a DAI link contains
only codec DAIs by examining the non_legacy_dai_naming flag in each
DAI's component.
For now, we assume there is only one or a small set of valid PCM stream
parameters, so num_params == 1 is good enough. We also assume that the
same params are valid for all supported streams. params is set to the
subset of parameters common among all DAIs, and then the existing code
automatically chooses the highest quality of the remaining values when
the link is brought up.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200305051143.60691-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
114 lines
3.8 KiB
ReStructuredText
114 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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.params = &dsp_codec_params,
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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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.params = &dsp_codec_params,
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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 "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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