Merge branch 'topic/doc' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/restize-docs

1 files changed, 103 insertions, 0 deletions

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..704a648
--- /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.