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+Dynamic PCM
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+===========
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+
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+1. Description
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+==============
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+
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+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
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+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
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+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
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+drivers that expose several ALSA PCMs and can route to multiple DAIs.
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+
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+The DPCM runtime routing is determined by the ALSA mixer settings in the same
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+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
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+graph representing the DSP internal audio paths and uses the mixer settings to
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+determine the patch used by each ALSA PCM.
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+
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+DPCM re-uses all the existing component codec, platform and DAI drivers without
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+any modifications.
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+
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+
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+Phone Audio System with SoC based DSP
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+-------------------------------------
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+
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+Consider the following phone audio subsystem. This will be used in this
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+document for all examples :-
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+
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+| Front End PCMs | SoC DSP | Back End DAIs | Audio devices |
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+
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+ *************
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+PCM0 <------------> * * <----DAI0-----> Codec Headset
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+ * *
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+PCM1 <------------> * * <----DAI1-----> Codec Speakers
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+ * DSP *
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+PCM2 <------------> * * <----DAI2-----> MODEM
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+ * *
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+PCM3 <------------> * * <----DAI3-----> BT
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+ * *
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+ * * <----DAI4-----> DMIC
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+ * *
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+ * * <----DAI5-----> FM
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+ *************
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+
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+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
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+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
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+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
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+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
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+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
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+
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+
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+
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+Example - DPCM Switching playback from DAI0 to DAI1
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+---------------------------------------------------
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+
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+Audio is being played to the Headset. After a while the user removes the headset
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+and audio continues playing on the speakers.
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+
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+Playback on PCM0 to Headset would look like :-
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+
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+ *************
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+PCM0 <============> * * <====DAI0=====> Codec Headset
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+ * *
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+PCM1 <------------> * * <----DAI1-----> Codec Speakers
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+ * DSP *
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+PCM2 <------------> * * <----DAI2-----> MODEM
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+ * *
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+PCM3 <------------> * * <----DAI3-----> BT
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+ * *
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+ * * <----DAI4-----> DMIC
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+ * *
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+ * * <----DAI5-----> FM
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+ *************
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+
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+The headset is removed from the jack by user so the speakers must now be used :-
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+
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+ *************
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+PCM0 <============> * * <----DAI0-----> Codec Headset
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+ * *
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+PCM1 <------------> * * <====DAI1=====> Codec Speakers
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+ * DSP *
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+PCM2 <------------> * * <----DAI2-----> MODEM
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+ * *
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+PCM3 <------------> * * <----DAI3-----> BT
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+ * *
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+ * * <----DAI4-----> DMIC
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+ * *
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+ * * <----DAI5-----> FM
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+ *************
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+
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+The audio driver processes this as follows :-
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+
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+ 1) Machine driver receives Jack removal event.
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+
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+ 2) Machine driver OR audio HAL disables the Headset path.
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+
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+ 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
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+ for headset since the path is now disabled.
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+
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+ 4) Machine driver or audio HAL enables the speaker path.
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+
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+ 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
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+ trigger(start) for DAI1 Speakers since the path is enabled.
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+
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+In this example, the machine driver or userspace audio HAL can alter the routing
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+and then DPCM will take care of managing the DAI PCM operations to either bring
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+the link up or down. Audio playback does not stop during this transition.
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+
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+
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+
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+DPCM machine driver
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+===================
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+
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+The DPCM enabled ASoC machine driver is similar to normal machine drivers
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+except that we also have to :-
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+
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+ 1) Define the FE and BE DAI links.
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+
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+ 2) Define any FE/BE PCM operations.
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+
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+ 3) Define widget graph connections.
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+
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+
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+1 FE and BE DAI links
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+---------------------
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+
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+| Front End PCMs | SoC DSP | Back End DAIs | Audio devices |
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+
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+ *************
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+PCM0 <------------> * * <----DAI0-----> Codec Headset
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+ * *
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+PCM1 <------------> * * <----DAI1-----> Codec Speakers
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+ * DSP *
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+PCM2 <------------> * * <----DAI2-----> MODEM
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+ * *
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+PCM3 <------------> * * <----DAI3-----> BT
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+ * *
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+ * * <----DAI4-----> DMIC
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+ * *
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+ * * <----DAI5-----> FM
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+ *************
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+
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+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
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+FE DAI links are defined as follows :-
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+
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+static struct snd_soc_dai_link machine_dais[] = {
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+ {
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+ .name = "PCM0 System",
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+ .stream_name = "System Playback",
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+ .cpu_dai_name = "System Pin",
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+ .platform_name = "dsp-audio",
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+ .codec_name = "snd-soc-dummy",
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+ .codec_dai_name = "snd-soc-dummy-dai",
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+ .dynamic = 1,
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+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
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+ .dpcm_playback = 1,
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+ },
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+ .....< other FE and BE DAI links here >
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+};
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+
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+This FE DAI link is pretty similar to a regular DAI link except that we also
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+set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream
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+directions should also be set with the "dpcm_playback" and "dpcm_capture"
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+flags. There is also an option to specify the ordering of the trigger call for
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+each FE. This allows the ASoC core to trigger the DSP before or after the other
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+components (as some DSPs have strong requirements for the ordering DAI/DSP
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+start and stop sequences).
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+
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+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
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+dynamic and will change depending on runtime config.
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+
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+The BE DAIs are configured as follows :-
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+
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+static struct snd_soc_dai_link machine_dais[] = {
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+ .....< FE DAI links here >
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+ {
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+ .name = "Codec Headset",
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+ .cpu_dai_name = "ssp-dai.0",
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+ .platform_name = "snd-soc-dummy",
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+ .no_pcm = 1,
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+ .codec_name = "rt5640.0-001c",
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+ .codec_dai_name = "rt5640-aif1",
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+ .ignore_suspend = 1,
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+ .ignore_pmdown_time = 1,
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+ .be_hw_params_fixup = hswult_ssp0_fixup,
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+ .ops = &haswell_ops,
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+ .dpcm_playback = 1,
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+ .dpcm_capture = 1,
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+ },
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+ .....< other BE DAI links here >
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+};
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+
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+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
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+the "no_pcm" flag to mark it has a BE and sets flags for supported stream
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+directions using "dpcm_playback" and "dpcm_capture" above.
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+
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+The BE has also flags set for ignoring suspend and PM down time. This allows
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+the BE to work in a hostless mode where the host CPU is not transferring data
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+like a BT phone call :-
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+
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+ *************
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+PCM0 <------------> * * <----DAI0-----> Codec Headset
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+ * *
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+PCM1 <------------> * * <----DAI1-----> Codec Speakers
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+ * DSP *
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+PCM2 <------------> * * <====DAI2=====> MODEM
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+ * *
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+PCM3 <------------> * * <====DAI3=====> BT
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+ * *
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+ * * <----DAI4-----> DMIC
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+ * *
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+ * * <----DAI5-----> FM
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+ *************
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+
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+This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
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+still in operation.
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+
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+A BE DAI link can also set the codec to a dummy device if the code is a device
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+that is managed externally.
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+
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+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
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+DSP firmware.
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+
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+
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+2 FE/BE PCM operations
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+----------------------
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+
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+The BE above also exports some PCM operations and a "fixup" callback. The fixup
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+callback is used by the machine driver to (re)configure the DAI based upon the
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+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
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+
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+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
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+DAI0. This means all FE hw_params have to be fixed in the machine driver for
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+DAI0 so that the DAI is running at desired configuration regardless of the FE
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+configuration.
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+
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+static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
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+ struct snd_pcm_hw_params *params)
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+{
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+ struct snd_interval *rate = hw_param_interval(params,
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+ SNDRV_PCM_HW_PARAM_RATE);
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+ struct snd_interval *channels = hw_param_interval(params,
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+ SNDRV_PCM_HW_PARAM_CHANNELS);
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+
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+ /* The DSP will covert the FE rate to 48k, stereo */
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+ rate->min = rate->max = 48000;
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+ channels->min = channels->max = 2;
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+
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+ /* set DAI0 to 16 bit */
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+ snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
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+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
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+ SNDRV_PCM_FORMAT_S16_LE);
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+ return 0;
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+}
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+
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+The other PCM operation are the same as for regular DAI links. Use as necessary.
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+
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+
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+3 Widget graph connections
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+--------------------------
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+
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+The BE DAI links will normally be connected to the graph at initialisation time
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+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
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+has to be set explicitly in the driver :-
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+
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+/* BE for codec Headset - DAI0 is dummy and managed by DSP FW */
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+{"DAI0 CODEC IN", NULL, "AIF1 Capture"},
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+{"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
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+
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+
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+Writing a DPCM DSP driver
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+=========================
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+
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+The DPCM DSP driver looks much like a standard platform class ASoC driver
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+combined with elements from a codec class driver. A DSP platform driver must
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+implement :-
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+
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+ 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
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+
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+ 2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
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+
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+ 3) DAPM widgets from DSP graph.
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+
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+ 4) Mixers for gains, routing, etc.
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+
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+ 5) DMA configuration.
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+
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+ 6) BE AIF widgets.
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+
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+Items 6 is important for routing the audio outside of the DSP. AIF need to be
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+defined for each BE and each stream direction. e.g for BE DAI0 above we would
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+have :-
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+
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+SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
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+SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
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+
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+The BE AIF are used to connect the DSP graph to the graphs for the other
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+component drivers (e.g. codec graph).
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+
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+
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+Hostless PCM streams
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+====================
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+
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+A hostless PCM stream is a stream that is not routed through the host CPU. An
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+example of this would be a phone call from handset to modem.
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+
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+
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+ *************
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+PCM0 <------------> * * <----DAI0-----> Codec Headset
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+ * *
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+PCM1 <------------> * * <====DAI1=====> Codec Speakers/Mic
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+ * DSP *
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+PCM2 <------------> * * <====DAI2=====> MODEM
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+ * *
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+PCM3 <------------> * * <----DAI3-----> BT
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+ * *
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+ * * <----DAI4-----> DMIC
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+ * *
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+ * * <----DAI5-----> FM
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+ *************
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+
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+In this case the PCM data is routed via the DSP. The host CPU in this use case
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+is only used for control and can sleep during the runtime of the stream.
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+
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+The host can control the hostless link either by :-
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+
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+ 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link
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+ is enabled or disabled by the state of the DAPM graph. This usually means
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+ there is a mixer control that can be used to connect or disconnect the path
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+ between both DAIs.
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+
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+ 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
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+ graph. Control is then carried out by the FE as regular PCM operations.
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+ This method gives more control over the DAI links, but requires much more
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+ userspace code to control the link. Its recommended to use CODEC<->CODEC
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+ unless your HW needs more fine grained sequencing of the PCM ops.
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+
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+
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+CODEC <-> CODEC link
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+--------------------
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+
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+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
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+The machine driver sets some additional parameters to the DAI link i.e.
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+
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+static const struct snd_soc_pcm_stream dai_params = {
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+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
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+ .rate_min = 8000,
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+ .rate_max = 8000,
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+ .channels_min = 2,
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+ .channels_max = 2,
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+};
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+
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+static struct snd_soc_dai_link dais[] = {
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+ < ... more DAI links above ... >
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+ {
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+ .name = "MODEM",
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+ .stream_name = "MODEM",
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+ .cpu_dai_name = "dai2",
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+ .codec_dai_name = "modem-aif1",
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+ .codec_name = "modem",
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+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
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+ | SND_SOC_DAIFMT_CBM_CFM,
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+ .params = &dai_params,
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+ }
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+ < ... more DAI links here ... >
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+
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+These parameters are used to configure the DAI hw_params() when DAPM detects a
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+valid path and then calls the PCM operations to start the link. DAPM will also
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+call the appropriate PCM operations to disable the DAI when the path is no
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+longer valid.
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+
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+
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+Hostless FE
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+-----------
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+
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+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
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+This means creating a new FE that is connected with a virtual path to both
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+DAI links. The DAI links will be started when the FE PCM is started and stopped
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+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
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+this configuration.
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+
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+
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