ASoC: doc: ReSTize DPCM.txt

A simple conversion from a plain text file.
The file name was renamed to lower letters to align with others.

Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 files changed, 393 insertions, 0 deletions

diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst
new file mode 100644
index 000000000000..395e5a516282
--- /dev/null
+++ b/Documentation/sound/soc/dpcm.rst
@@ -0,0 +1,392 @@
+===========
+Dynamic PCM
+===========
+
+Description
+===========
+
+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
+drivers that expose several ALSA PCMs and can route to multiple DAIs.
+
+The DPCM runtime routing is determined by the ALSA mixer settings in the same
+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
+graph representing the DSP internal audio paths and uses the mixer settings to
+determine the patch used by each ALSA PCM.
+
+DPCM re-uses all the existing component codec, platform and DAI drivers without
+any modifications.
+
+
+Phone Audio System with SoC based DSP
+-------------------------------------
+
+Consider the following phone audio subsystem. This will be used in this
+document for all examples :-
+::
+
+  | Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+  
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
+
+
+
+Example - DPCM Switching playback from DAI0 to DAI1
+---------------------------------------------------
+
+Audio is being played to the Headset. After a while the user removes the headset
+and audio continues playing on the speakers.
+
+Playback on PCM0 to Headset would look like :-
+::
+
+                      *************
+  PCM0 <============> *           * <====DAI0=====> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+The headset is removed from the jack by user so the speakers must now be used :-
+::
+
+                      *************
+  PCM0 <============> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <====DAI1=====> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+The audio driver processes this as follows :-
+
+1. Machine driver receives Jack removal event.
+
+2. Machine driver OR audio HAL disables the Headset path.
+
+3. DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
+   for headset since the path is now disabled.
+
+4. Machine driver or audio HAL enables the speaker path.
+
+5. DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
+   trigger(start) for DAI1 Speakers since the path is enabled.
+
+In this example, the machine driver or userspace audio HAL can alter the routing
+and then DPCM will take care of managing the DAI PCM operations to either bring
+the link up or down. Audio playback does not stop during this transition.
+
+
+
+DPCM machine driver
+===================
+
+The DPCM enabled ASoC machine driver is similar to normal machine drivers
+except that we also have to :-
+
+1. Define the FE and BE DAI links.
+
+2. Define any FE/BE PCM operations.
+
+3. Define widget graph connections.
+
+
+FE and BE DAI links
+-------------------
+::
+
+  | Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+  
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
+FE DAI links are defined as follows :-
+::
+
+  static struct snd_soc_dai_link machine_dais[] = {
+	{
+		.name = "PCM0 System",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "dsp-audio",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dynamic = 1,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	.....< other FE and BE DAI links here >
+  };
+
+This FE DAI link is pretty similar to a regular DAI link except that we also
+set the DAI link to a DPCM FE with the ``dynamic = 1``. The supported FE stream
+directions should also be set with the ``dpcm_playback`` and ``dpcm_capture``
+flags. There is also an option to specify the ordering of the trigger call for
+each FE. This allows the ASoC core to trigger the DSP before or after the other
+components (as some DSPs have strong requirements for the ordering DAI/DSP
+start and stop sequences).
+
+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
+dynamic and will change depending on runtime config.
+
+The BE DAIs are configured as follows :-
+::
+
+  static struct snd_soc_dai_link machine_dais[] = {
+	.....< FE DAI links here >
+	{
+		.name = "Codec Headset",
+		.cpu_dai_name = "ssp-dai.0",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "rt5640.0-001c",
+		.codec_dai_name = "rt5640-aif1",
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = hswult_ssp0_fixup,
+		.ops = &haswell_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	.....< other BE DAI links here >
+  };
+
+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
+the ``no_pcm`` flag to mark it has a BE and sets flags for supported stream
+directions using ``dpcm_playback`` and ``dpcm_capture`` above.
+
+The BE has also flags set for ignoring suspend and PM down time. This allows
+the BE to work in a hostless mode where the host CPU is not transferring data
+like a BT phone call :-
+::
+
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <====DAI2=====> MODEM
+                      *           *
+  PCM3 <------------> *           * <====DAI3=====> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
+still in operation.
+
+A BE DAI link can also set the codec to a dummy device if the code is a device
+that is managed externally.
+
+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
+DSP firmware.
+
+
+FE/BE PCM operations
+--------------------
+
+The BE above also exports some PCM operations and a ``fixup`` callback. The fixup
+callback is used by the machine driver to (re)configure the DAI based upon the
+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
+
+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
+DAI0. This means all FE hw_params have to be fixed in the machine driver for
+DAI0 so that the DAI is running at desired configuration regardless of the FE
+configuration.
+::
+
+  static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+  {
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set DAI0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+  }
+
+The other PCM operation are the same as for regular DAI links. Use as necessary.
+
+
+Widget graph connections
+------------------------
+
+The BE DAI links will normally be connected to the graph at initialisation time
+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
+has to be set explicitly in the driver :-
+::
+
+  /* BE for codec Headset -  DAI0 is dummy and managed by DSP FW */
+  {"DAI0 CODEC IN", NULL, "AIF1 Capture"},
+  {"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
+
+
+Writing a DPCM DSP driver
+=========================
+
+The DPCM DSP driver looks much like a standard platform class ASoC driver
+combined with elements from a codec class driver. A DSP platform driver must
+implement :-
+
+1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
+
+2. DAPM graph showing DSP audio routing from FE DAIs to BEs.
+
+3. DAPM widgets from DSP graph.
+
+4. Mixers for gains, routing, etc.
+
+5. DMA configuration.
+
+6. BE AIF widgets.
+
+Items 6 is important for routing the audio outside of the DSP. AIF need to be
+defined for each BE and each stream direction. e.g for BE DAI0 above we would
+have :-
+::
+
+  SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+  SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+The BE AIF are used to connect the DSP graph to the graphs for the other
+component drivers (e.g. codec graph).
+
+
+Hostless PCM streams
+====================
+
+A hostless PCM stream is a stream that is not routed through the host CPU. An
+example of this would be a phone call from handset to modem.
+::
+
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <====DAI1=====> Codec Speakers/Mic
+                      *   DSP     *
+  PCM2 <------------> *           * <====DAI2=====> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+In this case the PCM data is routed via the DSP. The host CPU in this use case
+is only used for control and can sleep during the runtime of the stream.
+
+The host can control the hostless link either by :-
+
+ 1. Configuring the link as a CODEC <-> CODEC style link. In this case the link
+    is enabled or disabled by the state of the DAPM graph. This usually means
+    there is a mixer control that can be used to connect or disconnect the path
+    between both DAIs.
+
+ 2. Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
+    graph. Control is then carried out by the FE as regular PCM operations.
+    This method gives more control over the DAI links, but requires much more
+    userspace code to control the link. Its recommended to use CODEC<->CODEC
+    unless your HW needs more fine grained sequencing of the PCM ops.
+
+
+CODEC <-> CODEC link
+--------------------
+
+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
+The machine driver sets some additional parameters to the DAI link i.e.
+::
+
+  static const struct snd_soc_pcm_stream dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+  };
+
+  static struct snd_soc_dai_link dais[] = {
+	< ... more DAI links above ... >
+	{
+		.name = "MODEM",
+		.stream_name = "MODEM",
+		.cpu_dai_name = "dai2",
+		.codec_dai_name = "modem-aif1",
+		.codec_name = "modem",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dai_params,
+	}
+	< ... more DAI links here ... >
+
+These parameters are used to configure the DAI hw_params() when DAPM detects a
+valid path and then calls the PCM operations to start the link. DAPM will also
+call the appropriate PCM operations to disable the DAI when the path is no
+longer valid.
+
+
+Hostless FE
+-----------
+
+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
+This means creating a new FE that is connected with a virtual path to both
+DAI links. The DAI links will be started when the FE PCM is started and stopped
+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
+this configuration.
+
+
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 85ec51764e83..e142a0f25c5b 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -16,3 +16,4 @@ The documentation is spilt into the following sections:-
    pops-clicks
    clocking
    jack
+   dpcm