Updated documentation to provide more details
for codec-to-codec connection.
Signed-off-by: anish kumar <yesanishhere@xxxxxxxxx>
---
v2: Fixed the compilation error reported by Sphinx
Documentation/sound/soc/codec-to-codec.rst | 296 +++++++++++++--------
1 file changed, 190 insertions(+), 106 deletions(-)
diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
index 0418521b6e03..9d65fc74856a 100644
--- a/Documentation/sound/soc/codec-to-codec.rst
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -1,115 +1,199 @@
-==============================================
-Creating codec to codec dai link for ALSA dapm
-==============================================
+Codec-to-Codec Connections in ALSA
+====================================
-Mostly the flow of audio is always from CPU to codec so your system
-will look as below:
-::
+An ALSA-based audio system typically involves playback and capture
+functionalities, where users may require audio file playback through
+speakers or recording from microphones. However, certain systems
+necessitate audio data routing directly between components, such as FM
+radio to speakers, without CPU involvement. For such scenarios, ALSA
+provides a mechanism known as codec-to-codec connections, leveraging
+the Dynamic Audio Power Management (DAPM) framework to facilitate
+direct data transfers between codecs.
- --------- ---------
- | | dai | |
- CPU -------> codec
- | | | |
- --------- ---------
+Introduction
+------------
-In case your system looks as below:
-::
+In most audio systems, audio data flows from the CPU to the codec. In
+specific configurations, such as those involving Bluetooth codecs,
+audio can be transmitted directly between codecs without CPU
+intervention. ALSA supports both architectures, and for systems that
+do not involve the CPU, it utilizes codec-to-codec digital audio
+interface (DAI) connections. This document discusses the procedure
+for establishing codec-to-codec DAI links to enable such
+functionalities.
+Audio Data Flow Paths
+----------------------
+
+In a typical configuration, audio flow can be visualized as follows:
+
+.. code-block:: text
+
+ --------- ---------
+ | | dai | |
+ CPU -------> codec
+ | | | |
+ --------- ---------
+
+In more intricate setups, the system may not involve the CPU but
+instead utilizes multiple codecs as shown below. For instance,
+Codec-2 acts as a cellular modem, while Codec-3 connects to a
+speaker. Audio data can be received by Codec-2 and transmitted to
+Codec-3 without CPU intervention, demonstrating the ideal conditions
+for establishing a codec-to-codec DAI connection.
+
+.. code-block:: text
+
+ ---------
+ | |
+ codec-1 <---cellular modem
+ | |
---------
- | |
- codec-2
- | |
- ---------
- |
- dai-2
- |
- ---------- ---------
- | | dai-1 | |
- CPU -------> codec-1
- | | | |
- ---------- ---------
- |
- dai-3
- |
- ---------
- | |
- codec-3
- | |
+ |
+ dai-1
+ ↓
+ ---------- ---------
+ | |cpu_dai | |
+ dummy CPU -------> codec-2
+ | | | |
+ ---------- ---------
+ |
+ dai-3
+ ↓
+ ---------
+ | |
+ codec-3 ---->speaker
+ | |
---------
-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,
- .c2c_params = &dsp_codec_params,
- .num_c2c_params = 1,
- },
- {
- .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,
- .c2c_params = &dsp_codec_params,
- .num_c2c_params = 1,
- },
-
-Above code snippet is motivated from sound/soc/samsung/speyside.c.
-
-Note the "c2c_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
+Creating Codec-to-Codec Connections in ALSA
+----------------------------------------------
+
+To create a codec-to-codec DAI in ALSA, a ``snd_soc_dai_link`` must be
+added to the machine driver before registering the sound card.
+During this registration, the core checks for the presence of
+``c2c_params`` within the ``snd_soc_dai_link``, determining whether
+to classify the DAI link as codec-to-codec.
+
+While establishing the PCM node, the ALSA core inspects this
+parameter. Instead of generating a user-space PCM node, it creates
+an internal PCM node utilized by kernel drivers. Consequently,
+running ``cat /proc/asound/pcm`` will yield no visible PCM nodes.
+
+After this setup, the ALSA core invokes the DAPM core to connect a
+single ``cpu_dai`` with both ``codec_dais``. Boot-up logs will
+display messages similar to:
+
+.. code-block:: bash
+
+ ASoC: registered pcm #0 codec2codec(Playback Codec)
+ multicodec <-> cpu_dai mapping ok
+ connected DAI link Dummy-CPU:cpu_dai -> codec-1:dai_1
+ connected DAI link Dummy-CPU:cpu_dai -> codec-2:dai_2
+
+To trigger this DAI link, a control interface is established by the
+DAPM core during internal DAI creation. This interface links to
+the ``snd_soc_dai_link_event`` function, which is invoked when a
+path connects in the DAPM core. A mixer must be created to trigger
+the connection, prompting the DAPM core to evaluate path
+connections and call the ``snd_soc_dai_link_event`` callback with
+relevant events.
+
+It is important to note that not all operations defined in
+``snd_soc_dai_ops`` are invoked as codec-to-codec connections offer
+limited control over DAI configuration. For greater control, a
+hostless configuration is recommended. The operations typically
+executed in codec-to-codec setups include startup, ``hw_params``,
+``hw_free``, digital mute, and shutdown from the
+``snd_soc_dai_ops`` structure.
+
+Code Changes for Codec-to-Codec
+----------------------------------
+
+The DAI link configuration in the machine file should resemble the
+following code snippet:
+
+.. code-block:: c
+
+ /*
+ * This PCM stream only supports 24-bit, 2 channels, and
+ * 48kHz 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,
+ };
+
+ static struct snd_soc_dai_link dai_links[] = {
+ {
+ .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,
+ .c2c_params = &dsp_codec_params,
+ .num_c2c_params = 1,
+ },
+ {
+ .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,
+ .c2c_params = &dsp_codec_params,
+ .num_c2c_params = 1,
+ },
+ };
+
+This snippet draws inspiration from the configuration found in
+``sound/soc/samsung/speyside.c``. The inclusion of the
+``c2c_params`` indicates to the DAPM core that the DAI link is a
+codec-to-codec connection.
+
+In the DAPM core, a route is established between the CPU DAI
+playback widget and the codec DAI capture widget for playback, with
+the reverse applying to the capture path. To trigger these routes,
+DAPM requires valid endpoints, which can be either sink or source
+widgets corresponding to the playback and capture paths,
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.
-
-A dai_link in a "simple-audio-card" will automatically be detected as
-codec to codec when all DAIs on the link belong to codec components.
-The dai_link will be initialized with the subset of stream parameters
-(channels, format, sample rate) supported by all DAIs on the link. Since
-there is no way to provide these parameters in the device tree, this is
-mostly useful for communication with simple fixed-function codecs, such
-as a Bluetooth controller or cellular modem.
+To activate this DAI link widget, a lightweight codec driver for
+the speaker amplifier can be implemented, following a similar
+strategy to that in ``wm8727.c``. This driver should set the
+necessary constraints for the device, even with minimal control
+requirements.
+
+It's crucial to append “Playback” and “Capture” suffixes to the
+respective CPU and codec DAI names for playback and capture, as
+the DAPM core links and powers these DAIs based on their naming
+conventions.
+
+In a codec-to-codec scenario involving multiple codecs (above
+bootup logs are for multicodec scenario), it is not feasible to
+control individual codecs using dummy kcontrols or DAPM widgets.
+This limitation arises because the CPU DAI is statically
+connected to all codecs. Consequently, when a path is enabled,
+the DAPM core does not verify all the widgets that may be linked
+to the mixer widget. It’s important to note that the mixer widget
+serves as the trigger for these paths.
+
+Simple-audio-card configuration
+----------------------------------
+A dai_link in a "simple-audio-card" will automatically be
+detected as codec-to-codec when all DAIs on the link belong to
+codec components. The dai_link will be initialized with the
+subset of stream parameters (channels, format, sample rate)
+supported by all DAIs on the link. Since there is no way to
+provide these parameters in the device tree, this is mostly useful
+for communication with simple fixed-function codecs, such as a
+Bluetooth controller or cellular modem.
--
2.39.3 (Apple Git-146)
[Index of Archives]
[Pulseaudio]
[Linux Audio Users]
[ALSA Devel]
[Fedora Desktop]
[Fedora SELinux]
[Big List of Linux Books]
[Yosemite News]
[KDE Users]
