| =========== |
| 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(¶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. |
| |
| |
| 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. |
| |
| |