Merge remote-tracking branches 'asoc/topic/davinci' and 'asoc/topic/doc' into asoc-next
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commit
a3d91900a6
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@ -43,7 +43,7 @@ mcbsp0: mcbsp@1d10000 {
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<0x00310000 0x1000>;
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reg-names = "mpu", "dat";
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interrupts = <97 98>;
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interrupts-names = "rx", "tx";
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interrupt-names = "rx", "tx";
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dmas = <&edma0 3 1
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&edma0 2 1>;
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dma-names = "tx", "rx";
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103
Documentation/sound/alsa/soc/codec_to_codec.txt
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103
Documentation/sound/alsa/soc/codec_to_codec.txt
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@ -0,0 +1,103 @@
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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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| | dai | |
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CPU -------> codec
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| | | |
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--------- ---------
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In case your system looks as below:
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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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---------- ---------
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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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* 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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Note that in current device tree there is no way to mark a dai_link
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as codec to codec. However, it may change in future.
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