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Documentation/sound/alsa/soc/DPCM.txt Normal file
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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.

46
Documentation/sound/alsa/soc/codec.txt
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@ -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 {

71
Documentation/sound/alsa/soc/dapm.txt
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@ -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

6
Documentation/sound/alsa/soc/machine.txt
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@ -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

19
Documentation/sound/alsa/soc/platform.txt
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@ -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.