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authorMark Brown <broonie@linaro.org>2013-10-24 10:24:05 (GMT)
committerMark Brown <broonie@linaro.org>2013-10-24 10:24:05 (GMT)
commit9f7a949fb909b017e054574c96c4ebb44f5ff3fa (patch)
tree58aa069ba011a9ef626f30a61cfcdd1ac520b1b8 /Documentation/sound
parent70e0db2f7434961c778466708c032e76775b7f1e (diff)
parent469b7bc4e6dbfdb173f0901f746e9277f6740ba7 (diff)
Merge remote-tracking branch 'asoc/topic/doc' into asoc-next
Diffstat (limited to 'Documentation/sound')
-rw-r--r--Documentation/sound/alsa/soc/DPCM.txt380
-rw-r--r--Documentation/sound/alsa/soc/codec.txt46
-rw-r--r--Documentation/sound/alsa/soc/dapm.txt71
-rw-r--r--Documentation/sound/alsa/soc/machine.txt6
-rw-r--r--Documentation/sound/alsa/soc/platform.txt19
5 files changed, 456 insertions, 66 deletions
diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt
new file mode 100644
index 0000000..aa8546f
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DPCM.txt
@@ -0,0 +1,380 @@
+Dynamic PCM
+===========
+
+1. 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.
+
+
+1 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 ignoreing 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.
+
+
+2 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.
+
+
+3 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 regualar 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/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index bce23a4..db5f9c9 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -1,22 +1,23 @@
-ASoC Codec Driver
-=================
+ASoC Codec Class Driver
+=======================
-The codec driver is generic and hardware independent code that configures the
-codec to provide audio capture and playback. It should contain no code that is
-specific to the target platform or machine. All platform and machine specific
-code should be added to the platform and machine drivers respectively.
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
-Each codec driver *must* provide the following features:-
+Each codec class driver *must* provide the following features:-
1) Codec DAI and PCM configuration
- 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
+ 2) Codec control IO - using RegMap API
3) Mixers and audio controls
4) Codec audio operations
+ 5) DAPM description.
+ 6) DAPM event handler.
Optionally, codec drivers can also provide:-
- 5) DAPM description.
- 6) DAPM event handler.
7) DAC Digital mute control.
Its probably best to use this guide in conjunction with the existing codec
@@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = {
2 - Codec control IO
--------------------
The codec can usually be controlled via an I2C or SPI style interface
-(AC97 combines control with data in the DAI). The codec drivers provide
-functions to read and write the codec registers along with supplying a
-register cache:-
-
- /* IO control data and register cache */
- void *control_data; /* codec control (i2c/3wire) data */
- void *reg_cache;
-
-Codec read/write should do any data formatting and call the hardware
-read write below to perform the IO. These functions are called by the
-core and ALSA when performing DAPM or changing the mixer:-
-
- unsigned int (*read)(struct snd_soc_codec *, unsigned int);
- int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
-
-Codec hardware IO functions - usually points to either the I2C, SPI or AC97
-read/write:-
-
- hw_write_t hw_write;
- hw_read_t hw_read;
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
3 - Mixers and audio controls
@@ -127,7 +111,7 @@ Defines a stereo enumerated control
4 - Codec Audio Operations
--------------------------
-The codec driver also supports the following ALSA operations:-
+The codec driver also supports the following ALSA PCM operations:-
/* SoC audio ops */
struct snd_soc_ops {
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 05bf5a0..7dfd88c 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -21,7 +21,7 @@ level power systems.
There are 4 power domains within DAPM
- 1. Codec domain - VREF, VMID (core codec and audio power)
+ 1. Codec bias domain - VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
@@ -63,14 +63,22 @@ Audio DAPM widgets fall into a number of types:-
o Line - Line Input/Output (and optional Jack)
o Speaker - Speaker
o Supply - Power or clock supply widget used by other widgets.
+ o Regulator - External regulator that supplies power to audio components.
+ o Clock - External clock that supplies clock to audio componnents.
+ o AIF IN - Audio Interface Input (with TDM slot mask).
+ o AIF OUT - Audio Interface Output (with TDM slot mask).
+ o Siggen - Signal Generator.
+ o DAI IN - Digital Audio Interface Input.
+ o DAI OUT - Digital Audio Interface Output.
+ o DAI Link - DAI Link between two DAI structures */
o Pre - Special PRE widget (exec before all others)
o Post - Special POST widget (exec after all others)
(Widgets are defined in include/sound/soc-dapm.h)
-Widgets are usually added in the codec driver and the machine driver. There are
-convenience macros defined in soc-dapm.h that can be used to quickly build a
-list of widgets of the codecs and machines DAPM widgets.
+Widgets can be added to the sound card by any of the component driver types.
+There are convenience macros defined in soc-dapm.h that can be used to quickly
+build a list of widgets of the codecs and machines DAPM widgets.
Most widgets have a name, register, shift and invert. Some widgets have extra
parameters for stream name and kcontrols.
@@ -80,11 +88,13 @@ parameters for stream name and kcontrols.
-------------------------
Stream Widgets relate to the stream power domain and only consist of ADCs
-(analog to digital converters) and DACs (digital to analog converters).
+(analog to digital converters), DACs (digital to analog converters),
+AIF IN and AIF OUT.
Stream widgets have the following format:-
SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
NOTE: the stream name must match the corresponding stream name in your codec
snd_soc_codec_dai.
@@ -94,6 +104,11 @@ e.g. stream widgets for HiFi playback and capture
SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
+e.g. stream widgets for AIF
+
+SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+
2.2 Path Domain Widgets
-----------------------
@@ -121,12 +136,14 @@ If you dont want the mixer elements prefixed with the name of the mixer widget,
you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
as for SND_SOC_DAPM_MIXER.
-2.3 Platform/Machine domain Widgets
------------------------------------
+
+2.3 Machine domain Widgets
+--------------------------
Machine widgets are different from codec widgets in that they don't have a
codec register bit associated with them. A machine widget is assigned to each
-machine audio component (non codec) that can be independently powered. e.g.
+machine audio component (non codec or DSP) that can be independently
+powered. e.g.
o Speaker Amp
o Microphone Bias
@@ -146,12 +163,12 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
-2.4 Codec Domain
-----------------
+2.4 Codec (BIAS) Domain
+-----------------------
-The codec power domain has no widgets and is handled by the codecs DAPM event
-handler. This handler is called when the codec powerstate is changed wrt to any
-stream event or by kernel PM events.
+The codec bias power domain has no widgets and is handled by the codecs DAPM
+event handler. This handler is called when the codec powerstate is changed wrt
+to any stream event or by kernel PM events.
2.5 Virtual Widgets
@@ -169,15 +186,16 @@ After all the widgets have been defined, they can then be added to the DAPM
subsystem individually with a call to snd_soc_dapm_new_control().
-3. Codec Widget Interconnections
-================================
+3. Codec/DSP Widget Interconnections
+====================================
-Widgets are connected to each other within the codec and machine by audio paths
-(called interconnections). Each interconnection must be defined in order to
-create a map of all audio paths between widgets.
+Widgets are connected to each other within the codec, platform and machine by
+audio paths (called interconnections). Each interconnection must be defined in
+order to create a map of all audio paths between widgets.
-This is easiest with a diagram of the codec (and schematic of the machine audio
-system), as it requires joining widgets together via their audio signal paths.
+This is easiest with a diagram of the codec or DSP (and schematic of the machine
+audio system), as it requires joining widgets together via their audio signal
+paths.
e.g., from the WM8731 output mixer (wm8731.c)
@@ -247,16 +265,9 @@ machine and includes the codec. e.g.
o Mic Jack
o Codec Pins
-When a codec pin is NC it can be marked as not used with a call to
-
-snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
-
-The last argument is 0 for inactive and 1 for active. This way the pin and its
-input widget will never be powered up and consume power.
-
-This also applies to machine widgets. e.g. if a headphone is connected to a
-jack then the jack can be marked active. If the headphone is removed, then
-the headphone jack can be marked inactive.
+Endpoints are added to the DAPM graph so that their usage can be determined in
+order to save power. e.g. NC codecs pins will be switched OFF, unconnected
+jacks can also be switched OFF.
5 DAPM Widget Events
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index d50c14d..74056db 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -1,8 +1,10 @@
ASoC Machine Driver
===================
-The ASoC machine (or board) driver is the code that glues together the platform
-and codec drivers.
+The ASoC machine (or board) driver is the code that glues together all the
+component drivers (e.g. codecs, platforms and DAIs). It also describes the
+relationships between each componnent which include audio paths, GPIOs,
+interrupts, clocking, jacks and voltage regulators.
The machine driver can contain codec and platform specific code. It registers
the audio subsystem with the kernel as a platform device and is represented by
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index d57efad..3a08a2c 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -1,9 +1,9 @@
ASoC Platform Driver
====================
-An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
-and control. The platform drivers only target the SoC CPU and must have no board
-specific code.
+An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
+drivers and DSP drivers. The platform drivers only target the SoC CPU and must
+have no board specific code.
Audio DMA
=========
@@ -64,3 +64,16 @@ Each SoC DAI driver must provide the following features:-
5) Suspend and resume (optional)
Please see codec.txt for a description of items 1 - 4.
+
+
+SoC DSP Drivers
+===============
+
+Each SoC DSP driver usually supplies the following features :-
+
+ 1) DAPM graph
+ 2) Mixer controls
+ 3) DMA IO to/from DSP buffers (if applicable)
+ 4) Definition of DSP front end (FE) PCM devices.
+
+Please see DPCM.txt for a description of item 4.

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