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Codec to codec dai link description

Message ID CABCoZhB6F9J0677swtrHdp=Yg_v6b=C9Ph5KhWqaDN=3UjwqkA@mail.gmail.com (mailing list archive)
State Accepted
Commit 452a256898e7ca88115aa02d3851e67994ce3e19
Headers show

Commit Message

anish kumar Oct. 24, 2016, 4:03 a.m. UTC
Signed-off-by: anish kumar <yesanishhere@gmail.com>
---
 Documentation/sound/alsa/soc/codec_to_codec.txt | 103 ++++++++++++++++++++++++
 1 file changed, 103 insertions(+)
 create mode 100644 Documentation/sound/alsa/soc/codec_to_codec.txt

Comments

Charles Keepax Oct. 24, 2016, 8:13 a.m. UTC | #1
On Sun, Oct 23, 2016 at 09:03:53PM -0700, anish kumar wrote:
> Signed-off-by: anish kumar <yesanishhere@gmail.com>
> ---

Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>

Thanks,
Charles
Mark Brown Oct. 26, 2016, 10:31 a.m. UTC | #2
On Sun, Oct 23, 2016 at 09:03:53PM -0700, anish kumar wrote:
> Signed-off-by: anish kumar <yesanishhere@gmail.com>
> ---

Please submit patches using subject lines reflecting the style for the
subsystem.  This makes it easier for people to identify relevant
patches.  Look at what existing commits in the area you're changing are
doing and make sure your subject lines visually resemble what they're
doing.
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Patch

diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt
b/Documentation/sound/alsa/soc/codec_to_codec.txt
new file mode 100644
index 0000000..61c9cae
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec_to_codec.txt
@@ -0,0 +1,103 @@ 
+Creating codec to codec dai link for ALSA dapm
+===================================================
+
+Mostly the flow of audio is always from CPU to codec so your system
+will look as below:
+
+ ---------          ---------
+|         |  dai   |         |
+    CPU    ------->    codec
+|         |        |         |
+ ---------          ---------
+
+In case your system looks as below:
+                     ---------
+                    |         |
+                      codec-2
+                    |         |
+                     ---------
+                         |
+                       dai-2
+                         |
+ ----------          ---------
+|          |  dai-1 |         |
+    CPU     ------->  codec-1
+|          |        |         |
+ ----------          ---------
+                         |
+                       dai-3
+                         |
+                     ---------
+                    |         |
+                      codec-3
+                    |         |
+                     ---------
+
+Suppose codec-2 is a bluetooth chip and codec-3 is connected to
+a speaker and you have a below scenario:
+codec-2 will receive the audio data and the user wants to play that
+audio through codec-3 without involving the CPU.This
+aforementioned case is the ideal case when codec to codec
+connection should be used.
+
+Your dai_link should appear as below in your machine
+file:
+
+/*
+ * this pcm stream only supports 24 bit, 2 channel and
+ * 48k sampling rate.
+ */
+static const struct snd_soc_pcm_stream dsp_codec_params = {
+        .formats = SNDRV_PCM_FMTBIT_S24_LE,
+        .rate_min = 48000,
+        .rate_max = 48000,
+        .channels_min = 2,
+        .channels_max = 2,
+};
+
+{
+    .name = "CPU-DSP",
+    .stream_name = "CPU-DSP",
+    .cpu_dai_name = "samsung-i2s.0",
+    .codec_name = "codec-2,
+    .codec_dai_name = "codec-2-dai_name",
+    .platform_name = "samsung-i2s.0",
+    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+            | SND_SOC_DAIFMT_CBM_CFM,
+    .ignore_suspend = 1,
+    .params = &dsp_codec_params,
+},
+{
+    .name = "DSP-CODEC",
+    .stream_name = "DSP-CODEC",
+    .cpu_dai_name = "wm0010-sdi2",
+    .codec_name = "codec-3,
+    .codec_dai_name = "codec-3-dai_name",
+    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+            | SND_SOC_DAIFMT_CBM_CFM,
+    .ignore_suspend = 1,
+    .params = &dsp_codec_params,
+},
+
+Above code snippet is motivated from sound/soc/samsung/speyside.c.
+
+Note the "params" callback which lets the dapm know that this
+dai_link is a codec to codec connection.
+
+In dapm core a route is created between cpu_dai playback widget
+and codec_dai capture widget for playback path and vice-versa is
+true for capture path. In order for this aforementioned route to get
+triggered, DAPM needs to find a valid endpoint which could be either
+a sink or source widget corresponding to playback and capture path
+respectively.
+
+In order to trigger this dai_link widget, a thin codec driver for
+the speaker amp can be created as demonstrated in wm8727.c file, it
+sets appropriate constraints for the device even if it needs no control.
+
+Make sure to name your corresponding cpu and codec playback and capture
+dai names ending with "Playback" and "Capture" respectively as dapm core
+will link and power those dais based on the name.
+
+Note that in current device tree there is no way to mark a dai_link
+as codec to codec. However, it may change in future.