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Signed-off-by: anish kumar --- 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.