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Signed-off-by: anish kumar --- 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 | 296 +++++++++++++-------- 1 file changed, 190 insertions(+), 106 deletions(-) diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sou= nd/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 @@ -=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D -Creating codec to codec dai link for ALSA dapm -=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D +Codec-to-Codec Connections in ALSA +=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D =20 -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. =20 - --------- --------- - | | dai | | - CPU -------> codec - | | | | - --------- --------- +Introduction +------------ =20 -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. =20 +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 + =E2=86=93 + ---------- --------- + | |cpu_dai | | + dummy CPU -------> codec-2 + | | | | + ---------- --------- + | + dai-3 + =E2=86=93 + --------- + | | + codec-3 ---->speaker + | | --------- =20 -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 =3D { - .formats =3D SNDRV_PCM_FMTBIT_S24_LE, - .rate_min =3D 48000, - .rate_max =3D 48000, - .channels_min =3D 2, - .channels_max =3D 2, - }; - - { - .name =3D "CPU-DSP", - .stream_name =3D "CPU-DSP", - .cpu_dai_name =3D "samsung-i2s.0", - .codec_name =3D "codec-2, - .codec_dai_name =3D "codec-2-dai_name", - .platform_name =3D "samsung-i2s.0", - .dai_fmt =3D SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, - .ignore_suspend =3D 1, - .c2c_params =3D &dsp_codec_params, - .num_c2c_params =3D 1, - }, - { - .name =3D "DSP-CODEC", - .stream_name =3D "DSP-CODEC", - .cpu_dai_name =3D "wm0010-sdi2", - .codec_name =3D "codec-3, - .codec_dai_name =3D "codec-3-dai_name", - .dai_fmt =3D SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, - .ignore_suspend =3D 1, - .c2c_params =3D &dsp_codec_params, - .num_c2c_params =3D 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 =3D { + .formats =3D SNDRV_PCM_FMTBIT_S24_LE, + .rate_min =3D 48000, + .rate_max =3D 48000, + .channels_min =3D 2, + .channels_max =3D 2, + }; + + static struct snd_soc_dai_link dai_links[] =3D { + { + .name =3D "CPU-DSP", + .stream_name =3D "CPU-DSP", + .cpu_dai_name =3D "samsung-i2s.0", + .codec_name =3D "codec-2", + .codec_dai_name =3D "codec-2-dai_name", + .platform_name =3D "samsung-i2s.0", + .dai_fmt =3D SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend =3D 1, + .c2c_params =3D &dsp_codec_params, + .num_c2c_params =3D 1, + }, + { + .name =3D "DSP-CODEC", + .stream_name =3D "DSP-CODEC", + .cpu_dai_name =3D "wm0010-sdi2", + .codec_name =3D "codec-3", + .codec_dai_name =3D "codec-3-dai_name", + .dai_fmt =3D SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend =3D 1, + .c2c_params =3D &dsp_codec_params, + .num_c2c_params =3D 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. =20 -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 =E2=80=9CPlayback=E2=80=9D and =E2=80=9CCapture=E2= =80=9D 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=E2=80=99s 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. --=20 2.39.3 (Apple Git-146)