Updated documentation to provide more details
for codec-to-codec connection especially around
the scenarios and DAPM core details for C2C
creation.
Signed-off-by: anish kumar <yesanishhere@xxxxxxxxx>
---
v4: sent wrong patch so corrected that in this.
v3: took care of comments from Charles Keepax and
as advised modified some details.
v2: Fixed the compilation error reported by Sphinx
Documentation/sound/soc/codec-to-codec.rst | 296 +++++++++++++--------
1 file changed, 190 insertions(+), 106 deletions(-)
Documentation/sound/soc/codec-to-codec.rst | 307 ++++++++++++++-------
1 file changed, 208 insertions(+), 99 deletions(-)
diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
index 0418521b6e03..c0d6e831ae4b 100644
--- a/Documentation/sound/soc/codec-to-codec.rst
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -1,115 +1,224 @@
-==============================================
-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
+functionality, 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, ASoC(
+ALSA system on chip) provides a mechanism known as codec-to-codec (C2C)
+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. ASoC supports both architectures, and for systems that
+do not involve the CPU, it utilizes C2C digital audio
+interface (DAI) connections. This document discusses the procedure
+for establishing C2C DAI links to enable such functionality.
+
+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 C2C DAI connection.
+
+.. code-block:: text
---------
| |
- codec-2
+ | codec-1 <---cellular modem
| |
---------
|
- dai-2
- |
- ---------- ---------
- | | dai-1 | |
- CPU -------> codec-1
- | | | |
- ---------- ---------
+ dai-1
+ ↓
+ --------- ---------
+ | |cpu_dai | |
+ | CPU -------> codec-2 |
+ | | | |
+ --------- ---------
|
dai-3
- |
+ ↓
---------
| |
- codec-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
-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.
+In the diagram above, two kinds of use cases can be supported:
+
+ 1. Host music playback: CPU -> codec-2 -> codec-3 -> Speaker
+
+ When an application on the host plays audio, ASoC informs the DAPM
+ (Dynamic Audio Power Management) core that the main CPU DAI
+ (Digital Audio Interface) is now an active source. DAPM then parses
+ through the audio graph until it finds the speaker sink.
+
+ The act of playing audio triggers the following power-up sequence:
+
+ - The ``CPU -> codec-2`` DAI is activated.
+ - DAPM powers up the C2C DAI link between codec-2 and codec-3, as
+ it is part of the active audio path.
+
+ 2. Cellular call: codec-1 -> codec-2 -> codec-3 -> Speaker
+
+ In this case, the host is not involved at all. The modem acts as the
+ audio source, and DAPM powers up everything between it and the sink
+ (i.e., the speaker). This power-up sequence involves:
+
+ - The C2C DAI link between codec-1 and codec-2.
+ - The C2C DAI link between codec-2 and codec-3.
+
+ DAPM ensures that all necessary components in the audio path from the
+ modem to the speaker are powered up, enabling direct audio playback
+ from the modem without host intervention.
+
+
+Creating Codec-to-Codec Connections in ALSA
+-------------------------------------------
+
+To create a C2C 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 C2C.
+
+While establishing the PCM node, the ASoC 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.
+
+Boot-up logs will display message similar to:
+
+.. code-block:: text
+
+ ASoC: registered pcm #0 codec2codec(Playback Codec)
+
+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
+SND_SOC_DAPM_*_PMU and SND_SOC_DAPM_*_PMD events.
+
+It is important to note that not all operations defined in
+``snd_soc_dai_ops`` are invoked as C2C connections offer
+limited control over DAI configuration. The operations typically
+executed in C2C setups include startup, ``hw_params``, ``hw_free``,
+digital mute, and shutdown from the ``snd_soc_dai_ops`` struct.
+
+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,
+ },
+ };
+
+To better understand the configuration inspired by the setup found in
+``sound/soc/samsung/speyside.c``, here are several key points:
+
+1. The presence of ``c2c_params`` informs the DAPM core that the DAI link
+ represents a C2C connection.
+
+2. ``c2c_params`` can be an array, and ``num_c2c_params`` defines the size
+ of this array.
+
+3. If ``num_c2c_params`` is 1:
+
+ - The C2C DAI is configured with the provided ``snd_soc_pcm_stream``
+ parameters.
+
+4. If ``num_c2c_params`` is greater than 1:
+
+ - A kcontrol is created, allowing the user to select the index of the
+ ``c2c_params`` array to be used.
+
+This flexible approach enables dynamic configuration of C2C
+connections based on runtime requirements.
+
+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.
+
+To activate this DAI link widget, codec driver is required.
+If it doesn't exist, thin codec driver 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.
+
+
+Simple-audio-card configuration
+-------------------------------
+
+A dai_link in a "simple-audio-card" will automatically be
+detected as C2C 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)
