diff options
Diffstat (limited to 'Documentation/sound/alsa')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/Audiophile-Usb.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/CMIPCI.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/compress_offload.txt | 6 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/DPCM.txt | 380 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/codec.txt | 46 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/dapm.txt | 73 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/machine.txt | 6 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/platform.txt | 19 |
9 files changed, 463 insertions, 73 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 95731a08f257..b8dd0df76952 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -616,7 +616,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. As default, snd-dummy drivers doesn't allocate the real buffers but either ignores read/write or mmap a single dummy page to all - buffer pages, in order to save the resouces. If your apps need + buffer pages, in order to save the resources. If your apps need the read/ written buffer data to be consistent, pass fake_buffer=0 option. diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 654dd3b694a8..e7a5ed4dcae8 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -232,7 +232,7 @@ The parameter can be given: # modprobe snd-usb-audio index=1 device_setup=0x09 * Or while configuring the modules options in your modules configuration file - (tipically a .conf file in /etc/modprobe.d/ directory: + (typically a .conf file in /etc/modprobe.d/ directory: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt index 16935c8561f7..4e36e6e809ca 100644 --- a/Documentation/sound/alsa/CMIPCI.txt +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -87,7 +87,7 @@ with 4 channels, and use the interleaved 4 channel data. -There are some control switchs affecting to the speaker connections: +There are some control switches affecting to the speaker connections: "Line-In Mode" - an enum control to change the behavior of line-in jack. Either "Line-In", "Rear Output" or "Bass Output" can diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index fd74ff26376e..630c492c3dc2 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -217,12 +217,12 @@ Not supported: would be enabled with ALSA kcontrols. - Audio policy/resource management. This API does not provide any - hooks to query the utilization of the audio DSP, nor any premption + hooks to query the utilization of the audio DSP, nor any preemption mechanisms. -- No notion of underun/overrun. Since the bytes written are compressed +- No notion of underrun/overrun. Since the bytes written are compressed in nature and data written/read doesn't translate directly to - rendered output in time, this does not deal with underrun/overun and + rendered output in time, this does not deal with underrun/overrun and maybe dealt in user-library Credits: diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt new file mode 100644 index 000000000000..0110180b7ac6 --- /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 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. + + +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(¶ms->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 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/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index bce23a4a7875..db5f9c9ae149 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 05bf5a0eee41..6faab4880006 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. @@ -30,7 +30,7 @@ There are 4 power domains within DAPM machine driver and responds to asynchronous events e.g when HP are inserted - 3. Path domain - audio susbsystem signal paths + 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. @@ -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 components. + 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 d50c14df3411..74056dba52be 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 d57efad37e0a..3a08a2c9150c 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. |