diff options
Diffstat (limited to 'Documentation/sound/soc')
| -rw-r--r-- | Documentation/sound/soc/clocking.rst | 12 | ||||
| -rw-r--r-- | Documentation/sound/soc/codec-to-codec.rst | 4 | ||||
| -rw-r--r-- | Documentation/sound/soc/dapm-graph.svg | 375 | ||||
| -rw-r--r-- | Documentation/sound/soc/dapm.rst | 170 | ||||
| -rw-r--r-- | Documentation/sound/soc/dpcm.rst | 32 | ||||
| -rw-r--r-- | Documentation/sound/soc/index.rst | 1 | ||||
| -rw-r--r-- | Documentation/sound/soc/machine.rst | 26 | ||||
| -rw-r--r-- | Documentation/sound/soc/usb.rst | 482 |
8 files changed, 1029 insertions, 73 deletions
diff --git a/Documentation/sound/soc/clocking.rst b/Documentation/sound/soc/clocking.rst index 32122d6877a3..25d016ea8b65 100644 --- a/Documentation/sound/soc/clocking.rst +++ b/Documentation/sound/soc/clocking.rst @@ -42,5 +42,17 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. +ASoC provided clock APIs +------------------------ +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_sysclk +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_clkdiv + +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_pll + +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_bclk_ratio diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst index 0418521b6e03..973c147d9d82 100644 --- a/Documentation/sound/soc/codec-to-codec.rst +++ b/Documentation/sound/soc/codec-to-codec.rst @@ -68,7 +68,7 @@ file: .codec_dai_name = "codec-2-dai_name", .platform_name = "samsung-i2s.0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ignore_suspend = 1, .c2c_params = &dsp_codec_params, .num_c2c_params = 1, @@ -80,7 +80,7 @@ file: .codec_name = "codec-3, .codec_dai_name = "codec-3-dai_name", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ignore_suspend = 1, .c2c_params = &dsp_codec_params, .num_c2c_params = 1, diff --git a/Documentation/sound/soc/dapm-graph.svg b/Documentation/sound/soc/dapm-graph.svg new file mode 100644 index 000000000000..d783672db815 --- /dev/null +++ b/Documentation/sound/soc/dapm-graph.svg @@ -0,0 +1,375 @@ +<?xml version="1.0" encoding="UTF-8" standalone="no"?> +<!DOCTYPE svg PUBLIC "-//W3C//DTD SVG 1.1//EN" + "http://www.w3.org/Graphics/SVG/1.1/DTD/svg11.dtd"> +<!-- Generated by graphviz version 2.43.0 (0) + --> +<!-- Title: G Pages: 1 --> +<svg width="900pt" height="630pt" + viewBox="0.00 0.00 900.00 630.00" xmlns="http://www.w3.org/2000/svg" xmlns:xlink="http://www.w3.org/1999/xlink"> +<g id="graph0" class="graph" transform="scale(1 1) rotate(0) translate(4 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501.5,-583"/> +<text text-anchor="middle" x="471" y="-567.8" font-family="sans-serif" font-size="14.00">MICR</text> +<text text-anchor="middle" x="471" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text> +</g> +<!-- cs42l51.0-004a_Mic Preamp Right --> +<g id="node21" class="node"> +<title>cs42l51.0-004a_Mic Preamp Right</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="550.5,-509 417.5,-509 417.5,-471 550.5,-471 550.5,-509"/> +<text text-anchor="middle" x="484" y="-493.8" font-family="sans-serif" font-size="14.00">Mic Preamp Right</text> +<text text-anchor="middle" x="484" y="-478.8" font-family="sans-serif" font-size="14.00">[mixer]</text> +</g> +<!-- cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right --> +<g id="edge13" class="edge"> +<title>cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right</title> +<path fill="none" stroke="black" d="M474.28,-544.83C475.67,-537.13 477.32,-527.97 478.87,-519.42"/> +<polygon fill="black" stroke="black" points="482.34,-519.88 480.68,-509.41 475.45,-518.63 482.34,-519.88"/> +</g> +<!-- cs42l51.0-004a_Mic Bias --> +<g id="node19" class="node"> +<title>cs42l51.0-004a_Mic Bias</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="409.5,-583 338.5,-583 338.5,-545 409.5,-545 409.5,-583"/> +<text text-anchor="middle" x="374" y="-567.8" font-family="sans-serif" font-size="14.00">Mic Bias</text> +<text text-anchor="middle" x="374" y="-552.8" font-family="sans-serif" font-size="14.00">[supply]</text> +</g> +<!-- cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL --> +<g id="edge11" class="edge"> +<title>cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL</title> +<path fill="none" stroke="black" d="M372.74,-544.83C372.2,-537.13 371.57,-527.97 370.97,-519.42"/> +<polygon fill="black" stroke="black" points="374.46,-519.15 370.28,-509.41 367.48,-519.63 374.46,-519.15"/> +</g> +<!-- cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA --> +<g id="edge10" class="edge"> +<title>cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA</title> +<path fill="none" stroke="black" d="M804.92,-470.99C804.27,-460.3 803.44,-446.5 802.71,-434.37"/> +<polygon fill="black" stroke="black" points="806.2,-434.03 802.1,-424.26 799.21,-434.45 806.2,-434.03"/> +</g> +<!-- cs42l51.0-004a_Right PGA --> +<g id="node27" class="node"> +<title>cs42l51.0-004a_Right PGA</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="688.5,-424 605.5,-424 605.5,-386 688.5,-386 688.5,-424"/> +<text text-anchor="middle" x="647" y="-408.8" font-family="sans-serif" font-size="14.00">Right PGA</text> +<text text-anchor="middle" x="647" y="-393.8" font-family="sans-serif" font-size="14.00">[pga]</text> +</g> +<!-- cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA --> +<g id="edge19" class="edge"> +<title>cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA</title> +<path fill="none" stroke="black" d="M643.87,-470.99C644.38,-460.3 645.05,-446.5 645.63,-434.37"/> +<polygon fill="black" stroke="black" points="649.13,-434.42 646.12,-424.26 642.14,-434.08 649.13,-434.42"/> +</g> +<!-- cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux --> +<g id="edge5" class="edge"> +<title>cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux</title> +<path fill="none" stroke="black" d="M560,-237.83C560,-230.13 560,-220.97 560,-212.42"/> +<polygon fill="black" stroke="black" points="563.5,-212.41 560,-202.41 556.5,-212.41 563.5,-212.41"/> +</g> +<!-- cs42l51.0-004a_Right ADC --> +<g id="node25" class="node"> +<title>cs42l51.0-004a_Right ADC</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="697,-350 613,-350 613,-312 697,-312 697,-350"/> +<text text-anchor="middle" x="655" y="-334.8" font-family="sans-serif" font-size="14.00">Right ADC</text> +<text text-anchor="middle" x="655" y="-319.8" font-family="sans-serif" font-size="14.00">[adc]</text> +</g> +<!-- cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture --> +<g id="edge3" class="edge"> +<title>cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture</title> +<path fill="none" stroke="black" d="M655,-311.83C655,-304.13 655,-294.97 655,-286.42"/> +<polygon fill="black" stroke="black" points="658.5,-286.41 655,-276.41 651.5,-286.41 658.5,-286.41"/> +</g> +<!-- cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR --> +<g id="edge7" class="edge"> +<title>cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR</title> +<path fill="none" stroke="black" d="M608,-89.83C608,-82.13 608,-72.97 608,-64.42"/> +<polygon fill="black" stroke="black" points="611.5,-64.41 608,-54.41 604.5,-64.41 611.5,-64.41"/> +</g> +<!-- cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC --> +<g id="edge17" class="edge"> +<title>cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC</title> +<path fill="none" stroke="black" d="M649.02,-385.83C649.87,-378.13 650.89,-368.97 651.84,-360.42"/> +<polygon fill="black" stroke="black" points="655.33,-360.74 652.95,-350.41 648.37,-359.97 655.33,-360.74"/> +</g> +<!-- hdmi-audio-codec.1.auto_TX --> +<g id="node30" class="node"> +<title>hdmi-audio-codec.1.auto_TX</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="281.5,-509 210.5,-509 210.5,-471 281.5,-471 281.5,-509"/> +<text text-anchor="middle" x="246" y="-493.8" font-family="sans-serif" font-size="14.00">TX</text> +<text text-anchor="middle" x="246" y="-478.8" font-family="sans-serif" font-size="14.00">[output]</text> +</g> +<!-- hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX --> +<g id="edge22" class="edge"> +<title>hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX</title> +<path fill="none" stroke="black" d="M254.22,-544.83C253.05,-537.13 251.65,-527.97 250.34,-519.42"/> +<polygon fill="black" stroke="black" points="253.78,-518.77 248.81,-509.41 246.86,-519.83 253.78,-518.77"/> +</g> +<!-- hdmi-audio-codec.1.auto_RX --> +<g id="node29" class="node"> +<title>hdmi-audio-codec.1.auto_RX</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="189.5,-583 118.5,-583 118.5,-545 189.5,-545 189.5,-583"/> +<text text-anchor="middle" x="154" y="-567.8" font-family="sans-serif" font-size="14.00">RX</text> +<text text-anchor="middle" x="154" y="-552.8" font-family="sans-serif" font-size="14.00">[output]</text> +</g> +<!-- hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture --> +<g id="edge20" class="edge"> +<title>hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture</title> +<path fill="none" stroke="black" d="M154.25,-544.83C154.36,-537.13 154.49,-527.97 154.61,-519.42"/> +<polygon fill="black" stroke="black" points="158.1,-519.46 154.74,-509.41 151.11,-519.36 158.1,-519.46"/> +</g> +</g> +</svg> diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index c3154ce6e1b2..73a42d5a9f30 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -7,8 +7,8 @@ Description Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices to use the minimum amount of power within the audio -subsystem at all times. It is independent of other kernel PM and as -such, can easily co-exist with the other PM systems. +subsystem at all times. It is independent of other kernel power +management frameworks and, as such, can easily co-exist with them. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or @@ -16,11 +16,32 @@ recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire -audio subsystem, this includes internal codec power blocks and machine -level power systems. +DAPM is based on two basic elements, called widgets and routes: -There are 4 power domains within DAPM + * a **widget** is every part of the audio hardware that can be enabled by + software when in use and disabled to save power when not in use + * a **route** is an interconnection between widgets that exists when sound + can flow from one widget to the other + +All DAPM power switching decisions are made automatically by consulting an +audio routing graph. This graph is specific to each sound card and spans +the whole sound card, so some DAPM routes connect two widgets belonging to +different components (e.g. the LINE OUT pin of a CODEC and the input pin of +an amplifier). + +The graph for the STM32MP1-DK1 sound card is shown in picture: + +.. kernel-figure:: dapm-graph.svg + :alt: Example DAPM graph + :align: center + +You can also generate compatible graph for your sound card using +`tools/sound/dapm-graph` utility. + +DAPM power domains +================== + +There are 4 power domains within DAPM: Codec bias domain VREF, VMID (core codec and audio power) @@ -47,17 +68,11 @@ Stream domain Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. -All DAPM power switching decisions are made automatically by consulting an audio -routing map of the whole machine. This map is specific to each machine and -consists of the interconnections between every audio component (including -internal codec components). All audio components that effect power are called -widgets hereafter. - DAPM Widgets ============ -Audio DAPM widgets fall into a number of types:- +Audio DAPM widgets fall into a number of types: Mixer Mixes several analog signals into a single analog signal. @@ -141,14 +156,14 @@ Stream Widgets relate to the stream power domain and only consist of ADCs (analog to digital converters), DACs (digital to analog converters), AIF IN and AIF OUT. -Stream widgets have the following format:- +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. +snd_soc_dai_driver. e.g. stream widgets for HiFi playback and capture :: @@ -167,7 +182,7 @@ Path Domain Widgets ------------------- Path domain widgets have a ability to control or affect the audio signal or -audio paths within the audio subsystem. They have the following form:- +audio paths within the audio subsystem. They have the following form: :: SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -207,7 +222,7 @@ powered. e.g. A machine widget can have an optional call back. e.g. Jack connector widget for an external Mic that enables Mic Bias -when the Mic is inserted:-:: +when the Mic is inserted:: static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) { @@ -221,7 +236,7 @@ when the Mic is inserted:-:: Codec (BIAS) Domain ------------------- -The codec bias power domain has no widgets and is handled by the codecs DAPM +The codec bias power domain has no widgets and is handled by the codec DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -229,17 +244,58 @@ to any stream event or by kernel PM events. Virtual Widgets --------------- -Sometimes widgets exist in the codec or machine audio map that don't have any +Sometimes widgets exist in the codec or machine audio graph that don't have any corresponding soft power control. In this case it is necessary to create a virtual widget - a widget with no control bits e.g. :: SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -This can be used to merge to signal paths together in software. +This can be used to merge two signal paths together in software. + +Registering DAPM controls +========================= + +In many cases the DAPM widgets are implemented statically in a ``static +const struct snd_soc_dapm_widget`` array in a codec driver, and simply +declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the +``struct snd_soc_component_driver``. + +Similarly, routes connecting them are implemented statically in a ``static +const struct snd_soc_dapm_route`` array and declared via the +``dapm_routes`` and ``num_dapm_routes`` fields of the same struct. + +With the above declared, the driver registration will take care of +populating them:: + + static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + ... + }; + + /* Target, Path, Source */ + static const struct snd_soc_dapm_route wm2000_audio_map[] = { + { "SPKN", NULL, "ANC Engine" }, + { "SPKP", NULL, "ANC Engine" }, + ... + }; -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(). + static const struct snd_soc_component_driver soc_component_dev_wm2000 = { + ... + .dapm_widgets = wm2000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets), + .dapm_routes = wm2000_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), + ... + }; + +In more complex cases the list of DAPM widgets and/or routes can be only +known at probe time. This happens for example when a driver supports +different models having a different set of features. In those cases +separate widgets and routes arrays implementing the case-specific features +can be registered programmatically by calling snd_soc_dapm_new_controls() +and snd_soc_dapm_add_routes(). Codec/DSP Widget Interconnections @@ -247,31 +303,29 @@ Codec/DSP Widget Interconnections 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. +order to create a graph of all audio paths between widgets. 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) - -The WM8731 output mixer has 3 inputs (sources) +For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources): 1. Line Bypass Input 2. DAC (HiFi playback) 3. Mic Sidetone Input -Each input in this example has a kcontrol associated with it (defined in example -above) and is connected to the output mixer via its kcontrol name. We can now -connect the destination widget (wrt audio signal) with its source widgets. -:: +Each input in this example has a kcontrol associated with it (defined in +the example above) and is connected to the output mixer via its kcontrol +name. We can now connect the destination widget (wrt audio signal) with its +source widgets. :: /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -So we have :- +So we have: * Destination Widget <=== Path Name <=== Source Widget, or * Sink, Path, Source, or @@ -280,12 +334,11 @@ So we have :- When there is no path name connecting widgets (e.g. a direct connection) we pass NULL for the path name. -Interconnections are created with a call to:- -:: +Interconnections are created with a call to:: snd_soc_dapm_connect_input(codec, sink, path, source); -Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +Finally, snd_soc_dapm_new_widgets() must be called after all widgets and interconnections have been registered with the core. This causes the core to scan the codec and machine so that the internal DAPM state matches the physical state of the machine. @@ -326,35 +379,44 @@ jacks can also be switched OFF. DAPM Widget Events ================== -Some widgets can register their interest with the DAPM core in PM events. -e.g. A Speaker with an amplifier registers a widget so the amplifier can be -powered only when the spk is in use. -:: +Widgets needing to implement a more complex behaviour than what DAPM can do +can set a custom "event handler" by setting a function pointer. An example +is a power supply needing to enable a GPIO:: - /* turn speaker amplifier on/off depending on use */ - static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); - return 0; + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(gpio_pa, true); + else + gpiod_set_value_cansleep(gpio_pa, false); + + return 0; } - /* corgi machine dapm widgets */ - static const struct snd_soc_dapm_widget wm8731_dapm_widgets = - SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + static const struct snd_soc_dapm_widget st_widgets[] = { + ... + SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, + sof_es8316_speaker_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + }; -Please see soc-dapm.h for all other widgets that support events. +See soc-dapm.h for all other widgets that support events. Event types ----------- -The following event types are supported by event widgets. -:: +The following event types are supported by event widgets:: /* dapm event types */ - #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ - #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ - #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ - #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ - #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ - #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ + #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ + #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ + #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ + #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ + #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ + #define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ + #define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) + #define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU) diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst index 2d7ad1d91504..7b6aeab3c207 100644 --- a/Documentation/sound/soc/dpcm.rst +++ b/Documentation/sound/soc/dpcm.rst @@ -147,25 +147,25 @@ 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 :- :: + SND_SOC_DAILINK_DEFS(pcm0, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_PLATFORM("dsp-audio"))); + 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", + SND_SOC_DAILINK_REG(pcm0), .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 +set the DAI link to a DPCM FE with the ``dynamic = 1``. +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). @@ -176,28 +176,26 @@ dynamic and will change depending on runtime config. The BE DAIs are configured as follows :- :: + SND_SOC_DAILINK_DEFS(headset, + DAILINK_COMP_ARRAY(COMP_CPU("ssp-dai.0")), + DAILINK_COMP_ARRAY(COMP_CODEC("rt5640.0-001c", "rt5640-aif1"))); + 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", + SND_SOC_DAILINK_REG(headset), .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 ``no_pcm`` flag to mark it has a BE. 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 @@ -367,7 +365,7 @@ The machine driver sets some additional parameters to the DAI link i.e. .codec_dai_name = "modem-aif1", .codec_name = "modem", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .c2c_params = &dai_params, .num_c2c_params = 1, } diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst index e57df2dab2fd..8bed8f8f48da 100644 --- a/Documentation/sound/soc/index.rst +++ b/Documentation/sound/soc/index.rst @@ -18,3 +18,4 @@ The documentation is spilt into the following sections:- jack dpcm codec-to-codec + usb diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst index 515c9444deaf..1828f5edca3e 100644 --- a/Documentation/sound/soc/machine.rst +++ b/Documentation/sound/soc/machine.rst @@ -71,6 +71,18 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. .ops = &corgi_ops, }; +In the above struct, dai’s are registered using names but you can pass +either dai name or device tree node but not both. Also, names used here +for cpu/codec/platform dais should be globally unique. + +Additionally below example macro can be used to register cpu, codec and +platform dai:: + + SND_SOC_DAILINK_DEFS(wm2200_cpu_dsp, + DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")), + DAILINK_COMP_ARRAY(COMP_CODEC("spi0.0", "wm0010-sdi1")), + DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0"))); + struct snd_soc_card then sets up the machine with its DAIs. e.g. :: @@ -81,6 +93,10 @@ struct snd_soc_card then sets up the machine with its DAIs. e.g. .num_links = 1, }; +Following this, ``devm_snd_soc_register_card`` can be used to register +the sound card. During the registration, the individual components +such as the codec, CPU, and platform are probed. If all these components +are successfully probed, the sound card gets registered. Machine Power Map ----------------- @@ -95,3 +111,13 @@ Machine Controls ---------------- Machine specific audio mixer controls can be added in the DAI init function. + + +Clocking Controls +----------------- + +As previously noted, clock configuration is handled within the machine driver. +For details on the clock APIs that the machine driver can utilize for +setup, please refer to Documentation/sound/soc/clocking.rst. However, the +callback needs to be registered by the CPU/Codec/Platform drivers to configure +the clocks that is needed for the corresponding device operation. diff --git a/Documentation/sound/soc/usb.rst b/Documentation/sound/soc/usb.rst new file mode 100644 index 000000000000..94c12f9d9dd1 --- /dev/null +++ b/Documentation/sound/soc/usb.rst @@ -0,0 +1,482 @@ +================ +ASoC USB support +================ + +Overview +======== +In order to leverage the existing USB sound device support in ALSA, the +ASoC USB APIs are introduced to allow the subsystems to exchange +configuration information. + +One potential use case would be to support USB audio offloading, which is +an implementation that allows for an alternate power-optimized path in the audio +subsystem to handle the transfer of audio data over the USB bus. This would +let the main processor to stay in lower power modes for longer duration. The +following is an example design of how the ASoC and ALSA pieces can be connected +together to achieve this: + +:: + + USB | ASoC + | _________________________ + | | ASoC Platform card | + | |_________________________| + | | | + | ___V____ ____V____ + | |ASoC BE | |ASoC FE | + | |DAI LNK | |DAI LNK | + | |________| |_________| + | ^ ^ ^ + | | |________| + | ___V____ | + | |SoC-USB | | + ________ ________ | | | + |USB SND |<--->|USBSND |<------------>|________| | + |(card.c)| |offld |<---------- | + |________| |________|___ | | | + ^ ^ | | | ____________V_________ + | | | | | |IPC | + __ V_______________V_____ | | | |______________________| + |USB SND (endpoint.c) | | | | ^ + |_________________________| | | | | + ^ | | | ___________V___________ + | | | |->|audio DSP | + ___________V_____________ | | |_______________________| + |XHCI HCD |<- | + |_________________________| | + + +SoC USB driver +============== +Structures +---------- +``struct snd_soc_usb`` + + - ``list``: list head for SND SoC struct list + - ``component``: reference to ASoC component + - ``connection_status_cb``: callback to notify connection events + - ``update_offload_route_info``: callback to fetch selected USB sound card/PCM + device + - ``priv_data``: driver data + +The snd_soc_usb structure can be referenced using the ASoC platform card +device, or a USB device (udev->dev). This is created by the ASoC BE DAI +link, and the USB sound entity will be able to pass information to the +ASoC BE DAI link using this structure. + +``struct snd_soc_usb_device`` + + - ``card_idx``: sound card index associated with USB sound device + - ``chip_idx``: USB sound chip array index + - ``cpcm_idx``: capture pcm device indexes associated with the USB sound device + - ``ppcm_idx``: playback pcm device indexes associated with the USB sound device + - ``num_playback``: number of playback streams + - ``num_capture``: number of capture streams + - ``list``: list head for the USB sound device list + +The struct snd_soc_usb_device is created by the USB sound offload driver. +This will carry basic parameters/limitations that will be used to +determine the possible offloading paths for this USB audio device. + +Functions +--------- +.. code-block:: rst + + int snd_soc_usb_find_supported_format(int card_idx, + struct snd_pcm_hw_params *params, int direction) +.. + + - ``card_idx``: the index into the USB sound chip array. + - ``params``: Requested PCM parameters from the USB DPCM BE DAI link + - ``direction``: capture or playback + +**snd_soc_usb_find_supported_format()** ensures that the requested audio profile +being requested by the external DSP is supported by the USB device. + +Returns 0 on success, and -EOPNOTSUPP on failure. + +.. code-block:: rst + + int snd_soc_usb_connect(struct device *usbdev, struct snd_soc_usb_device *sdev) +.. + + - ``usbdev``: the usb device that was discovered + - ``sdev``: capabilities of the device + +**snd_soc_usb_connect()** notifies the ASoC USB DCPM BE DAI link of a USB +audio device detection. This can be utilized in the BE DAI +driver to keep track of available USB audio devices. This is intended +to be called by the USB offload driver residing in USB SND. + +Returns 0 on success, negative error code on failure. + +.. code-block:: rst + + int snd_soc_usb_disconnect(struct device *usbdev, struct snd_soc_usb_device *sdev) +.. + + - ``usbdev``: the usb device that was removed + - ``sdev``: capabilities to free + +**snd_soc_usb_disconnect()** notifies the ASoC USB DCPM BE DAI link of a USB +audio device removal. This is intended to be called by the USB offload +driver that resides in USB SND. + +.. code-block:: rst + + void *snd_soc_usb_find_priv_data(struct device *usbdev) +.. + + - ``usbdev``: the usb device to reference to find private data + +**snd_soc_usb_find_priv_data()** fetches the private data saved to the SoC USB +device. + +Returns pointer to priv_data on success, NULL on failure. + +.. code-block:: rst + + int snd_soc_usb_setup_offload_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack) +.. + + - ``component``: ASoC component to add the jack + - ``jack``: jack component to populate + +**snd_soc_usb_setup_offload_jack()** is a helper to add a sound jack control to +the platform sound card. This will allow for consistent naming to be used on +designs that support USB audio offloading. Additionally, this will enable the +jack to notify of changes. + +Returns 0 on success, negative otherwise. + +.. code-block:: rst + + int snd_soc_usb_update_offload_route(struct device *dev, int card, int pcm, + int direction, enum snd_soc_usb_kctl path, + long *route) +.. + + - ``dev``: USB device to look up offload path mapping + - ``card``: USB sound card index + - ``pcm``: USB sound PCM device index + - ``direction``: direction to fetch offload routing information + - ``path``: kcontrol selector - pcm device or card index + - ``route``: mapping of sound card and pcm indexes for the offload path. This is + an array of two integers that will carry the card and pcm device indexes + in that specific order. This can be used as the array for the kcontrol + output. + +**snd_soc_usb_update_offload_route()** calls a registered callback to the USB BE DAI +link to fetch the information about the mapped ASoC devices for executing USB audio +offload for the device. ``route`` may be a pointer to a kcontrol value output array, +which carries values when the kcontrol is read. + +Returns 0 on success, negative otherwise. + +.. code-block:: rst + + struct snd_soc_usb *snd_soc_usb_allocate_port(struct snd_soc_component *component, + void *data); +.. + + - ``component``: DPCM BE DAI link component + - ``data``: private data + +**snd_soc_usb_allocate_port()** allocates a SoC USB device and populates standard +parameters that is used for further operations. + +Returns a pointer to struct soc_usb on success, negative on error. + +.. code-block:: rst + + void snd_soc_usb_free_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to free + +**snd_soc_usb_free_port()** frees a SoC USB device. + +.. code-block:: rst + + void snd_soc_usb_add_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to add + +**snd_soc_usb_add_port()** add an allocated SoC USB device to the SOC USB framework. +Once added, this device can be referenced by further operations. + +.. code-block:: rst + + void snd_soc_usb_remove_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to remove + +**snd_soc_usb_remove_port()** removes a SoC USB device from the SoC USB framework. +After removing a device, any SOC USB operations would not be able to reference the +device removed. + +How to Register to SoC USB +-------------------------- +The ASoC DPCM USB BE DAI link is the entity responsible for allocating and +registering the SoC USB device on the component bind. Likewise, it will +also be responsible for freeing the allocated resources. An example can +be shown below: + +.. code-block:: rst + + static int q6usb_component_probe(struct snd_soc_component *component) + { + ... + data->usb = snd_soc_usb_allocate_port(component, 1, &data->priv); + if (!data->usb) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + + ret = snd_soc_usb_add_port(usb); + if (ret < 0) { + dev_err(component->dev, "failed to add usb port\n"); + goto free_usb; + } + ... + } + + static void q6usb_component_remove(struct snd_soc_component *component) + { + ... + snd_soc_usb_remove_port(data->usb); + snd_soc_usb_free_port(data->usb); + } + + static const struct snd_soc_component_driver q6usb_dai_component = { + .probe = q6usb_component_probe, + .remove = q6usb_component_remove, + .name = "q6usb-dai-component", + ... + }; +.. + +BE DAI links can pass along vendor specific information as part of the +call to allocate the SoC USB device. This will allow any BE DAI link +parameters or settings to be accessed by the USB offload driver that +resides in USB SND. + +USB Audio Device Connection Flow +-------------------------------- +USB devices can be hotplugged into the USB ports at any point in time. +The BE DAI link should be aware of the current state of the physical USB +port, i.e. if there are any USB devices with audio interface(s) connected. +connection_status_cb() can be used to notify the BE DAI link of any change. + +This is called whenever there is a USB SND interface bind or remove event, +using snd_soc_usb_connect() or snd_soc_usb_disconnect(): + +.. code-block:: rst + + static void qc_usb_audio_offload_probe(struct snd_usb_audio *chip) + { + ... + snd_soc_usb_connect(usb_get_usb_backend(udev), sdev); + ... + } + + static void qc_usb_audio_offload_disconnect(struct snd_usb_audio *chip) + { + ... + snd_soc_usb_disconnect(usb_get_usb_backend(chip->dev), dev->sdev); + ... + } +.. + +In order to account for conditions where driver or device existence is +not guaranteed, USB SND exposes snd_usb_rediscover_devices() to resend the +connect events for any identified USB audio interfaces. Consider the +the following situation: + + **usb_audio_probe()** + | --> USB audio streams allocated and saved to usb_chip[] + | --> Propagate connect event to USB offload driver in USB SND + | --> **snd_soc_usb_connect()** exits as USB BE DAI link is not ready + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> The USB audio device connect event is missed + +To ensure connection events are not missed, **snd_usb_rediscover_devices()** +is executed when the SoC USB device is registered. Now, when the BE DAI +link component probe occurs, the following highlights the sequence: + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> SoC USB device added, and **snd_usb_rediscover_devices()** runs + + **snd_usb_rediscover_devices()** + | --> Traverses through usb_chip[] and for non-NULL entries issue + | **connection_status_cb()** + +In the case where the USB offload driver is unbound, while USB SND is ready, +the **snd_usb_rediscover_devices()** is called during module init. This allows +for the offloading path to also be enabled with the following flow: + + **usb_audio_probe()** + | --> USB audio streams allocated and saved to usb_chip[] + | --> Propagate connect event to USB offload driver in USB SND + | --> USB offload driver **NOT** ready! + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> No USB connect event due to missing USB offload driver + + USB offload driver probe + | --> **qc_usb_audio_offload_init()** + | --> Calls **snd_usb_rediscover_devices()** to notify of devices + +USB Offload Related Kcontrols +============================= +Details +------- +A set of kcontrols can be utilized by applications to help select the proper sound +devices to enable USB audio offloading. SoC USB exposes the get_offload_dev() +callback that designs can use to ensure that the proper indices are returned to the +application. + +Implementation +-------------- + +**Example:** + + **Sound Cards**: + + :: + + 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D + SM8250-MTP-WCD9380-WSA8810-VA-DMIC + 1 [Seri ]: USB-Audio - Plantronics Blackwire 3225 Seri + Plantronics Plantronics Blackwire + 3225 Seri at usb-xhci-hcd.1.auto-1.1, + full sp + 2 [C320M ]: USB-Audio - Plantronics C320-M + Plantronics Plantronics C320-M at usb-xhci-hcd.1.auto-1.2, full speed + + **PCM Devices**: + + :: + + card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 0: MultiMedia1 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 1: MultiMedia2 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 1: Seri [Plantronics Blackwire 3225 Seri], device 0: USB Audio [USB Audio] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 2: C320M [Plantronics C320-M], device 0: USB Audio [USB Audio] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + + **USB Sound Card** - card#1: + + :: + + USB Offload Playback Card Route PCM#0 -1 (range -1->32) + USB Offload Playback PCM Route PCM#0 -1 (range -1->255) + + **USB Sound Card** - card#2: + + :: + + USB Offload Playback Card Route PCM#0 0 (range -1->32) + USB Offload Playback PCM Route PCM#0 1 (range -1->255) + +The above example shows a scenario where the system has one ASoC platform card +(card#0) and two USB sound devices connected (card#1 and card#2). When reading +the available kcontrols for each USB audio device, the following kcontrols lists +the mapped offload card and pcm device indexes for the specific USB device: + + ``USB Offload Playback Card Route PCM#*`` + + ``USB Offload Playback PCM Route PCM#*`` + +The kcontrol is indexed, because a USB audio device could potentially have +several PCM devices. The above kcontrols are defined as: + + - ``USB Offload Playback Card Route PCM#`` **(R)**: Returns the ASoC platform sound + card index for a mapped offload path. The output **"0"** (card index) signifies + that there is an available offload path for the USB SND device through card#0. + If **"-1"** is seen, then no offload path is available for the USB SND device. + This kcontrol exists for each USB audio device that exists in the system, and + its expected to derive the current status of offload based on the output value + for the kcontrol along with the PCM route kcontrol. + + - ``USB Offload Playback PCM Route PCM#`` **(R)**: Returns the ASoC platform sound + PCM device index for a mapped offload path. The output **"1"** (PCM device index) + signifies that there is an available offload path for the USB SND device through + PCM device#0. If **"-1"** is seen, then no offload path is available for the USB\ + SND device. This kcontrol exists for each USB audio device that exists in the + system, and its expected to derive the current status of offload based on the + output value for this kcontrol, in addition to the card route kcontrol. + +USB Offload Playback Route Kcontrol +----------------------------------- +In order to allow for vendor specific implementations on audio offloading device +selection, the SoC USB layer exposes the following: + +.. code-block:: rst + + int (*update_offload_route_info)(struct snd_soc_component *component, + int card, int pcm, int direction, + enum snd_soc_usb_kctl path, + long *route) +.. + +These are specific for the **USB Offload Playback Card Route PCM#** and **USB +Offload PCM Route PCM#** kcontrols. + +When users issue get calls to the kcontrol, the registered SoC USB callbacks will +execute the registered function calls to the DPCM BE DAI link. + +**Callback Registration:** + +.. code-block:: rst + + static int q6usb_component_probe(struct snd_soc_component *component) + { + ... + usb = snd_soc_usb_allocate_port(component, 1, &data->priv); + if (IS_ERR(usb)) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + usb->update_offload_route_info = q6usb_get_offload_dev; + + ret = snd_soc_usb_add_port(usb); +.. + +Existing USB Sound Kcontrol +--------------------------- +With the introduction of USB offload support, the above USB offload kcontrol +will be added to the pre existing list of kcontrols identified by the USB sound +framework. These kcontrols are still the main controls that are used to +modify characteristics pertaining to the USB audio device. + + :: + + Number of controls: 9 + ctl type num name value + 0 INT 2 Capture Channel Map 0, 0 (range 0->36) + 1 INT 2 Playback Channel Map 0, 0 (range 0->36) + 2 BOOL 1 Headset Capture Switch On + 3 INT 1 Headset Capture Volume 10 (range 0->13) + 4 BOOL 1 Sidetone Playback Switch On + 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) + 6 BOOL 1 Headset Playback Switch On + 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) + 8 INT 1 USB Offload Playback Card Route PCM#0 0 (range -1->32) + 9 INT 1 USB Offload Playback PCM Route PCM#0 1 (range -1->255) + +Since USB audio device controls are handled over the USB control endpoint, use the +existing mechanisms present in the USB mixer to set parameters, such as volume. |
