From 2d4e31de5bb2b5fbdbcd8a3bfec0eae0bd4ca409 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Apr 2017 01:35:41 +0000 Subject: ASoC: add audio-graph-card document "Audio Graph Card" = "Simple Card" + "OF-graph" Signed-off-by: Kuninori Morimoto Reviewed-by: Rob Herring Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++++++++++++++ 1 file changed, 124 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt new file mode 100644 index 000000000000..bac4b1b1060f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -0,0 +1,124 @@ +Audio Graph Card: + +Audio Graph Card specifies audio DAI connections of SoC <-> codec. +It is based on common bindings for device graphs. +see ${LINUX}/Documentation/devicetree/bindings/graph.txt + +Basically, Audio Graph Card property is same as Simple Card. +see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt + +Below are same as Simple-Card. + +- label +- dai-format +- frame-master +- bitclock-master +- bitclock-inversion +- frame-inversion +- dai-tdm-slot-num +- dai-tdm-slot-width +- clocks / system-clock-frequency + +Required properties: + +- compatible : "audio-graph-card"; +- dais : list of CPU DAI port{s} + +Example: Single DAI case + + sound_card { + compatible = "audio-graph-card"; + + dais = <&cpu_port>; + }; + + dai-controller { + ... + cpu_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + }; + + audio-codec { + ... + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; + +Example: Multi DAI case + + sound-card { + compatible = "audio-graph-card"; + + label = "sound-card"; + + dais = <&cpu_port0 + &cpu_port1 + &cpu_port2>; + }; + + audio-codec@0 { + ... + port { + codec0_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint0>; + }; + }; + }; + + audio-codec@1 { + ... + port { + codec1_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; + }; + }; + + audio-codec@2 { + ... + port { + codec2_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint2>; + }; + }; + }; + + dai-controller { + ... + ports { + cpu_port0: port@0 { + cpu_endpoint0: endpoint { + remote-endpoint = <&codec0_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + cpu_port1: port@1 { + cpu_endpoint1: endpoint { + remote-endpoint = <&codec1_endpoint>; + + dai-format = "i2s"; + ... + }; + }; + cpu_port2: port@2 { + cpu_endpoint2: endpoint { + remote-endpoint = <&codec2_endpoint>; + + dai-format = "i2s"; + ... + }; + }; + }; + }; + -- cgit v1.2.3 From 0378bb966765e06fdb0e00d592a6fcfbb71afa3c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 26 Apr 2017 02:26:30 +0000 Subject: ASoC: rsnd: move "renesas,rsrc-card" to "simple-scu-audio-card" on Document "renesas,rsrc-card" is exchanged to "simple-scu-card". Let's update Document Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316e4c91..3332910a9a11 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- cgit v1.2.3 From 5b3889f80a79d9b73f73e19ba9a001c2d47ab47f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 26 Apr 2017 02:26:48 +0000 Subject: ASoC: rsnd: add missing clocks/clock-names on Document Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 3332910a9a11..13cb21d53b8b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -368,6 +368,10 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI +- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. +- clock-names : List of necessary clock names. + "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", + "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout -- cgit v1.2.3 From 55cfebfede61cedf60289a7bd20e93a1c83a39a9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 18 May 2017 01:44:27 +0000 Subject: ASoC: simple-scu-card: cleanup documentation simple-scu-card is almost same as simple-card. This is already explained in document. But simple-card and simple-scu-card both has same explanation for same property. This patch forward explantion to simple-card if possible to avoid duplication. This patch also cleanup DT binding example which is not good matching to simple-scu-card. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/simple-scu-card.txt | 65 ++++++++-------------- 1 file changed, 24 insertions(+), 41 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt index d6fe47ed09af..e894cef1d314 100644 --- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-scu-card.txt @@ -1,35 +1,29 @@ -ASoC simple SCU Sound Card +ASoC Simple SCU Sound Card -Simple-Card specifies audio DAI connections of SoC <-> codec. +Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM". +For example, you can use this driver if you want to exchange sampling rate convert, +Mixing, etc... Required properties: - compatible : "simple-scu-audio-card" "renesas,rsrc-card" - Optional properties: -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,cpu : CPU sub-node -- simple-audio-card,codec : CODEC sub-node +- simple-audio-card,name : see simple-audio-card.txt +- simple-audio-card,cpu : see simple-audio-card.txt +- simple-audio-card,codec : see simple-audio-card.txt Optional subnode properties: -- simple-audio-card,format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- simple-audio-card,frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- simple-audio-card,bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- simple-audio-card,bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- simple-audio-card,frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. +- simple-audio-card,format : see simple-audio-card.txt +- simple-audio-card,frame-master : see simple-audio-card.txt +- simple-audio-card,bitclock-master : see simple-audio-card.txt +- simple-audio-card,bitclock-inversion : see simple-audio-card.txt +- simple-audio-card,frame-inversion : see simple-audio-card.txt - simple-audio-card,convert-rate : platform specified sampling rate convert - simple-audio-card,convert-channels : platform specified converted channel size (2 - 8 ch) -- simple-audio-card,prefix : see audio-routing +- simple-audio-card,prefix : see routing - simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources. @@ -38,19 +32,11 @@ Optional subnode properties: Required CPU/CODEC subnodes properties: -- sound-dai : phandle and port of CPU/CODEC +- sound-dai : see simple-audio-card.txt Optional CPU/CODEC subnodes properties: -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). +- clocks / system-clock-frequency : see simple-audio-card.txt Example 1. Sampling Rate Covert @@ -59,11 +45,10 @@ sound { simple-audio-card,name = "rsnd-ak4643"; simple-audio-card,format = "left_j"; - simple-audio-card,format = "left_j"; simple-audio-card,bitclock-master = <&sndcodec>; simple-audio-card,frame-master = <&sndcodec>; - simple-audio-card,convert-rate = <48000>; /* see audio_clk_a */ + simple-audio-card,convert-rate = <48000>; simple-audio-card,prefix = "ak4642"; simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", @@ -79,20 +64,18 @@ sound { }; }; -Example 2. 2 CPU 1 Codec +Example 2. 2 CPU 1 Codec (Mixing) sound { - compatible = "renesas,rsrc-card"; - - card-name = "rsnd-ak4643"; - format = "left_j"; - bitclock-master = <&dpcmcpu>; - frame-master = <&dpcmcpu>; + compatible = "simple-scu-audio-card"; - convert-rate = <48000>; /* see audio_clk_a */ + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; - audio-prefix = "ak4642"; - audio-routing = "ak4642 Playback", "DAI0 Playback", + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", "ak4642 Playback", "DAI1 Playback"; dpcmcpu: cpu@0 { -- cgit v1.2.3 From cff7597a46454cf0ef0de7340dfb3f6bc0855777 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 18 May 2017 16:32:37 +0100 Subject: ASoC: cs35l35: Add DT handling for Inductor Need to specify the inductor size in nH. This is a required property. Signed-off-by: Brian Austin Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs35l35.txt | 3 +++ 1 file changed, 3 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt index 016b768bc722..77ee75c39233 100644 --- a/Documentation/devicetree/bindings/sound/cs35l35.txt +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -16,6 +16,9 @@ Required properties: (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for further information relating to interrupt properties) + - cirrus,boost-ind-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1200nH, 1500nH, and 2200nH. + Optional properties: - reset-gpios : gpio used to reset the amplifier -- cgit v1.2.3 From da23173d5c6558b7435e71a4ad947390a9012c6c Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Thu, 18 May 2017 17:19:51 +0200 Subject: ASoC: stm32: Document STM32 I2S bindings Add documentation of device tree bindings for STM32 SPI/I2S. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,stm32-i2s.txt | 62 ++++++++++++++++++++++ 1 file changed, 62 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt new file mode 100644 index 000000000000..4bda52042402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt @@ -0,0 +1,62 @@ +STMicroelectronics STM32 SPI/I2S Controller + +The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. +Only some SPI instances support I2S. + +Required properties: + - compatible: Must be "st,stm32h7-i2s" + - reg: Offset and length of the device's register set. + - interrupts: Must contain the interrupt line id. + - clocks: Must contain phandle and clock specifier pairs for each entry + in clock-names. + - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". + "i2sclk": clock which feeds the internal clock generator + "pclk": clock which feeds the peripheral bus interface + "x8k": I2S parent clock for sampling rates multiple of 8kHz. + "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. + - dmas: DMA specifiers for tx and rx dma. + See Documentation/devicetree/bindings/dma/stm32-dma.txt. + - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". + - pinctrl-names: should contain only value "default" + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + +Optional properties: + - resets: Reference to a reset controller asserting the reset controller + +The device node should contain one 'port' child node with one child 'endpoint' +node, according to the bindings defined in Documentation/devicetree/bindings/ +graph.txt. + +Example: +sound_card { + compatible = "audio-graph-card"; + dais = <&i2s2_port>; +}; + +i2s2: audio-controller@40003800 { + compatible = "st,stm32h7-i2s"; + reg = <0x40003800 0x400>; + interrupts = <36>; + clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + dmas = <&dmamux2 2 39 0x400 0x1>, + <&dmamux2 3 40 0x400 0x1>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s2>; + + i2s2_port: port@0 { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + format = "i2s"; + }; + }; +}; + +audio-codec { + codec_port: port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; +}; -- cgit v1.2.3 From be10ee2cd351818738097c782b4493e09c6d14f6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 18 May 2017 01:44:46 +0000 Subject: ASoC: add audio-graph-scu-card document Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/audio-graph-scu-card.txt | 72 ++++++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt new file mode 100644 index 000000000000..b2dd23fd2135 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -0,0 +1,72 @@ +Audio-Graph-SCU-Card: + +Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". + +It is based on common bindings for device graphs. +see ${LINUX}/Documentation/devicetree/bindings/graph.txt + +Basically, Audio-Graph-SCU-Card property is same as +Simple-Card / Simple-SCU-Card / Audio-Graph-Card. +see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt + ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt + ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt + +Below are same as Simple-Card / Audio-Graph-Card. + +- label +- dai-format +- frame-master +- bitclock-master +- bitclock-inversion +- frame-inversion +- dai-tdm-slot-num +- dai-tdm-slot-width +- clocks / system-clock-frequency + +Below are same as Simple-SCU-Card. + +- convert-rate +- convert-channels +- prefix +- routing + +Required properties: + +- compatible : "audio-graph-scu-card"; +- dais : list of CPU DAI port{s} + +Example + + sound_card { + compatible = "audio-graph-scu-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback"; + convert-rate = <48000>; + + dais = <&cpu_port>; + }; + + audio-codec { + ... + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; + + dai-controller { + ... + cpu_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + }; -- cgit v1.2.3 From f7a478178a8ea970abd34f7ab73e66c9119b1606 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Jun 2017 22:36:02 +0200 Subject: ALSA: doc: Update copy_user, copy_kernel and fill_silence PCM ops Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- .../sound/kernel-api/writing-an-alsa-driver.rst | 111 ++++++++++++++------- 1 file changed, 76 insertions(+), 35 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 95c5443eff38..58ffa3f5bda7 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -2080,8 +2080,8 @@ sleeping poll threads, etc. This callback is also atomic as default. -copy and silence callbacks -~~~~~~~~~~~~~~~~~~~~~~~~~~ +copy_user, copy_kernel and fill_silence ops +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ These callbacks are not mandatory, and can be omitted in most cases. These callbacks are used when the hardware buffer cannot be in the @@ -3532,8 +3532,9 @@ external hardware buffer in interrupts (or in tasklets, preferably). The first case works fine if the external hardware buffer is large enough. This method doesn't need any extra buffers and thus is more -effective. You need to define the ``copy`` and ``silence`` callbacks -for the data transfer. However, there is a drawback: it cannot be +effective. You need to define the ``copy_user`` and ``copy_kernel`` +callbacks for the data transfer, in addition to ``fill_silence`` +callback for playback. However, there is a drawback: it cannot be mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. The second case allows for mmap on the buffer, although you have to @@ -3545,30 +3546,34 @@ Another case is when the chip uses a PCI memory-map region for the buffer instead of the host memory. In this case, mmap is available only on certain architectures like the Intel one. In non-mmap mode, the data cannot be transferred as in the normal way. Thus you need to define the -``copy`` and ``silence`` callbacks as well, as in the cases above. The -examples are found in ``rme32.c`` and ``rme96.c``. +``copy_user``, ``copy_kernel`` and ``fill_silence`` callbacks as well, +as in the cases above. The examples are found in ``rme32.c`` and +``rme96.c``. -The implementation of the ``copy`` and ``silence`` callbacks depends -upon whether the hardware supports interleaved or non-interleaved -samples. The ``copy`` callback is defined like below, a bit -differently depending whether the direction is playback or capture: +The implementation of the ``copy_user``, ``copy_kernel`` and +``silence`` callbacks depends upon whether the hardware supports +interleaved or non-interleaved samples. The ``copy_user`` callback is +defined like below, a bit differently depending whether the direction +is playback or capture: :: - static int playback_copy(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count); - static int capture_copy(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count); + static int playback_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void __user *src, unsigned long count); + static int capture_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void __user *dst, unsigned long count); In the case of interleaved samples, the second argument (``channel``) is not used. The third argument (``pos``) points the current position -offset in frames. +offset in bytes. The meaning of the fourth argument is different between playback and capture. For playback, it holds the source data pointer, and for capture, it's the destination data pointer. -The last argument is the number of frames to be copied. +The last argument is the number of bytes to be copied. What you have to do in this callback is again different between playback and capture directions. In the playback case, you copy the given amount @@ -3578,8 +3583,7 @@ way, the copy would be like: :: - my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src, - frames_to_bytes(runtime, count)); + my_memcpy_from_user(my_buffer + pos, src, count); For the capture direction, you copy the given amount of data (``count``) at the specified offset (``pos``) on the hardware buffer to the @@ -3587,31 +3591,68 @@ specified pointer (``dst``). :: - my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos), - frames_to_bytes(runtime, count)); + my_memcpy_to_user(dst, my_buffer + pos, count); + +Here the functions are named as ``from_user`` and ``to_user`` because +it's the user-space buffer that is passed to these callbacks. That +is, the callback is supposed to copy from/to the user-space data +directly to/from the hardware buffer. -Note that both the position and the amount of data are given in frames. +Careful readers might notice that these callbacks receive the +arguments in bytes, not in frames like other callbacks. It's because +it would make coding easier like the examples above, and also it makes +easier to unify both the interleaved and non-interleaved cases, as +explained in the following. In the case of non-interleaved samples, the implementation will be a bit -more complicated. +more complicated. The callback is called for each channel, passed by +the second argument, so totally it's called for N-channels times per +transfer. + +The meaning of other arguments are almost same as the interleaved +case. The callback is supposed to copy the data from/to the given +user-space buffer, but only for the given channel. For the detailed +implementations, please check ``isa/gus/gus_pcm.c`` or +"pci/rme9652/rme9652.c" as examples. + +The above callbacks are the copy from/to the user-space buffer. There +are some cases where we want copy from/to the kernel-space buffer +instead. In such a case, ``copy_kernel`` callback is called. It'd +look like: + +:: + + static int playback_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void *src, unsigned long count); + static int capture_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void *dst, unsigned long count); + +As found easily, the only difference is that the buffer pointer is +without ``__user`` prefix; that is, a kernel-buffer pointer is passed +in the fourth argument. Correspondingly, the implementation would be +a version without the user-copy, such as: -You need to check the channel argument, and if it's -1, copy the whole -channels. Otherwise, you have to copy only the specified channel. Please -check ``isa/gus/gus_pcm.c`` as an example. +:: + + my_memcpy(my_buffer + pos, src, count); -The ``silence`` callback is also implemented in a similar way +Usually for the playback, another callback ``fill_silence`` is +defined. It's implemented in a similar way as the copy callbacks +above: :: static int silence(struct snd_pcm_substream *substream, int channel, - snd_pcm_uframes_t pos, snd_pcm_uframes_t count); + unsigned long pos, unsigned long count); -The meanings of arguments are the same as in the ``copy`` callback, -although there is no ``src/dst`` argument. In the case of interleaved -samples, the channel argument has no meaning, as well as on ``copy`` -callback. +The meanings of arguments are the same as in the ``copy_user`` and +``copy_kernel`` callbacks, although there is no buffer pointer +argument. In the case of interleaved samples, the channel argument has +no meaning, as well as on ``copy_*`` callbacks. -The role of ``silence`` callback is to set the given amount +The role of ``fill_silence`` callback is to set the given amount (``count``) of silence data at the specified offset (``pos``) on the hardware buffer. Suppose that the data format is signed (that is, the silent-data is 0), and the implementation using a memset-like function @@ -3619,11 +3660,11 @@ would be like: :: - my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0, - frames_to_bytes(runtime, count)); + my_memset(my_buffer + pos, 0, count); In the case of non-interleaved samples, again, the implementation -becomes a bit more complicated. See, for example, ``isa/gus/gus_pcm.c``. +becomes a bit more complicated, as it's called N-times per transfer +for each channel. See, for example, ``isa/gus/gus_pcm.c``. Non-Contiguous Buffers ---------------------- -- cgit v1.2.3 From 010f21b2ba0cd546d7489ca536a156b281faa3fb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 6 Jun 2017 02:35:43 +0000 Subject: ASoC: simple-scu-card: tidyup "Sampling Rate Conversion" "Sampling Rate Conversion" is better than "Sampling Rate Convert" Reported-by: James Cameron Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/simple-scu-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt index e894cef1d314..327d229a51b2 100644 --- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-scu-card.txt @@ -38,7 +38,7 @@ Optional CPU/CODEC subnodes properties: - clocks / system-clock-frequency : see simple-audio-card.txt -Example 1. Sampling Rate Covert +Example 1. Sampling Rate Conversion sound { compatible = "simple-scu-audio-card"; -- cgit v1.2.3 From 08862251476f899c4c4528e922df5854076b8661 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 6 Jun 2017 02:37:07 +0000 Subject: ASoC: audio-graph-scu-card: add missing MIX binding example Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/audio-graph-scu-card.txt | 47 +++++++++++++++++++++- 1 file changed, 46 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b2dd23fd2135..b63c5594bbb3 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -35,7 +35,7 @@ Required properties: - compatible : "audio-graph-scu-card"; - dais : list of CPU DAI port{s} -Example +Example 1. Sampling Rate Conversion sound_card { compatible = "audio-graph-scu-card"; @@ -70,3 +70,48 @@ Example }; }; }; + +Example 2. 2 CPU 1 Codec (Mixing) + + sound_card { + compatible = "audio-graph-scu-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback"; + convert-rate = <48000>; + + dais = <&cpu_port0 + &cpu_port1>; + }; + + audio-codec { + ... + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint0>; + }; + }; + }; + + dai-controller { + ... + ports { + cpu_port0: port { + cpu_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + cpu_port1: port { + cpu_endpoint1: endpoint { + dai-format = "left_j"; + ... + }; + }; + }; + }; -- cgit v1.2.3 From 2cfeaec0ec896bc0b8aad2de28a3de4572c7e4a1 Mon Sep 17 00:00:00 2001 From: Icenowy Zheng Date: Mon, 5 Jun 2017 21:27:21 +0800 Subject: ASoC: sun8i-codec-analog: add support for V3s SoC The V3s SoC features an analog codec with headphone support but without mic2 and linein. Add support for it. Signed-off-by: Icenowy Zheng Reviewed-by: Chen-Yu Tsai Acked-by: Rob Herring Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt | 1 + sound/soc/sunxi/sun8i-codec-analog.c | 9 +++++++++ 2 files changed, 10 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt index 779b735781ba..1b6e7c4e50ab 100644 --- a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt +++ b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt @@ -4,6 +4,7 @@ Required properties: - compatible: must be one of the following compatibles: - "allwinner,sun8i-a23-codec-analog" - "allwinner,sun8i-h3-codec-analog" + - "allwinner,sun8i-v3s-codec-analog" Required properties if not a sub-node of the PRCM node: - reg: must contain the registers location and length diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 29c446068151..485e79f292c4 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -810,6 +810,11 @@ static int sun8i_codec_analog_add_mixer(struct snd_soc_component *cmpnt, return 0; } +static const struct sun8i_codec_analog_quirks sun8i_v3s_quirks = { + .has_headphone = true, + .has_hmic = true, +}; + static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) { struct device *dev = cmpnt->dev; @@ -886,6 +891,10 @@ static const struct of_device_id sun8i_codec_analog_of_match[] = { .compatible = "allwinner,sun8i-h3-codec-analog", .data = &sun8i_h3_quirks, }, + { + .compatible = "allwinner,sun8i-v3s-codec-analog", + .data = &sun8i_v3s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun8i_codec_analog_of_match); -- cgit v1.2.3 From 8b2840b6daca728cecfa925b50bf638189e2fbca Mon Sep 17 00:00:00 2001 From: Icenowy Zheng Date: Mon, 5 Jun 2017 21:27:22 +0800 Subject: ASoC: sun4i-codec: Add support for V3s codec The codec in the V3s is similar to the one found on the A31. One key difference is the analog path controls are routed through the PRCM block. This is supported by the sun8i-codec-analog driver, and tied into this codec driver with the audio card's aux_dev. In addition, the V3s does not have LINEIN, LINEOUT, MBIAS and MIC2, MIC3, and the FIFO related registers are like H3. Signed-off-by: Icenowy Zheng Reviewed-by: Chen-Yu Tsai Acked-by: Rob Herring Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sun4i-codec.txt | 11 ++-- sound/soc/sunxi/sun4i-codec.c | 63 ++++++++++++++++++++++ 2 files changed, 70 insertions(+), 4 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 3863531d1e6d..2d4e10deb6f4 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -7,6 +7,7 @@ Required properties: - "allwinner,sun7i-a20-codec" - "allwinner,sun8i-a23-codec" - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" - reg: must contain the registers location and length - interrupts: must contain the codec interrupt - dmas: DMA channels for tx and rx dma. See the DMA client binding, @@ -25,6 +26,7 @@ Required properties for the following compatibles: - "allwinner,sun6i-a31-codec" - "allwinner,sun8i-a23-codec" - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" - resets: phandle to the reset control for this device - allwinner,audio-routing: A list of the connections between audio components. Each entry is a pair of strings, the first being the @@ -34,15 +36,15 @@ Required properties for the following compatibles: Audio pins on the SoC: "HP" "HPCOM" - "LINEIN" - "LINEOUT" (not on sun8i-a23) + "LINEIN" (not on sun8i-v3s) + "LINEOUT" (not on sun8i-a23 or sun8i-v3s) "MIC1" - "MIC2" + "MIC2" (not on sun8i-v3s) "MIC3" (sun6i-a31 only) Microphone biases from the SoC: "HBIAS" - "MBIAS" + "MBIAS" (not on sun8i-v3s) Board connectors: "Headphone" @@ -55,6 +57,7 @@ Required properties for the following compatibles: Required properties for the following compatibles: - "allwinner,sun8i-a23-codec" - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" - allwinner,codec-analog-controls: A phandle to the codec analog controls block in the PRCM. diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index c3aab10fa085..150069987c0c 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1339,6 +1339,44 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return card; }; +static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return ERR_PTR(-ENOMEM); + + aux_dev.codec_of_node = of_parse_phandle(dev->of_node, + "allwinner,codec-analog-controls", + 0); + if (!aux_dev.codec_of_node) { + dev_err(dev, "Can't find analog controls for codec.\n"); + return ERR_PTR(-EINVAL); + }; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return ERR_PTR(-ENOMEM); + + card->dev = dev; + card->name = "V3s Audio Codec"; + card->dapm_widgets = sun6i_codec_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); + card->dapm_routes = sun8i_codec_card_routes; + card->num_dapm_routes = ARRAY_SIZE(sun8i_codec_card_routes); + card->aux_dev = &aux_dev; + card->num_aux_devs = 1; + card->fully_routed = true; + + ret = snd_soc_of_parse_audio_routing(card, "allwinner,audio-routing"); + if (ret) + dev_warn(dev, "failed to parse audio-routing: %d\n", ret); + + return card; +}; + static const struct regmap_config sun4i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, @@ -1374,6 +1412,13 @@ static const struct regmap_config sun8i_h3_codec_regmap_config = { .max_register = SUN8I_H3_CODEC_ADC_DBG, }; +static const struct regmap_config sun8i_v3s_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN8I_H3_CODEC_ADC_DBG, +}; + struct sun4i_codec_quirks { const struct regmap_config *regmap_config; const struct snd_soc_codec_driver *codec; @@ -1437,6 +1482,20 @@ static const struct sun4i_codec_quirks sun8i_h3_codec_quirks = { .has_reset = true, }; +static const struct sun4i_codec_quirks sun8i_v3s_codec_quirks = { + .regmap_config = &sun8i_v3s_codec_regmap_config, + /* + * TODO The codec structure should be split out, like + * H3, when adding digital audio processing support. + */ + .codec = &sun8i_a23_codec_codec, + .create_card = sun8i_v3s_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN6I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN8I_H3_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN6I_CODEC_ADC_RXDATA, + .has_reset = true, +}; + static const struct of_device_id sun4i_codec_of_match[] = { { .compatible = "allwinner,sun4i-a10-codec", @@ -1458,6 +1517,10 @@ static const struct of_device_id sun4i_codec_of_match[] = { .compatible = "allwinner,sun8i-h3-codec", .data = &sun8i_h3_codec_quirks, }, + { + .compatible = "allwinner,sun8i-v3s-codec", + .data = &sun8i_v3s_codec_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); -- cgit v1.2.3 From 5f440c48c5d5a11f1892819c409e183b2056e4ba Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Thu, 1 Jun 2017 12:36:15 +0200 Subject: ASoC: rsnd: Document optional reset properties Document the optional properties for describing module resets, to support resetting these modules on R-Car Gen2 and Gen3. Note that the audio module has resets for the Serial Sound Interfaces only. Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 3 +++ 1 file changed, 3 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 13cb21d53b8b..7246bb268bf9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -379,6 +379,9 @@ Optional properties: - clock-frequency : for all audio_clkout0/1/2/3 - clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn is asynchronizes with lr-clock. +- resets : References to SSI resets. +- reset-names : List of valid reset names. + "ssi-all", "ssi.X" SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer -- cgit v1.2.3 From 63ddf5dc9cc77963a993921995d9e390095dc3d4 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 9 Jun 2017 19:09:44 +0200 Subject: ASoC: samsung: Odroid DT binding documentation corrections This patch removes unused and undocumented samsung,cpu-dai, samsung,codec-dai properties from the dts example and moves sub-nodes' description to a separate section. Suggested-by: Rob Herring Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/samsung,odroid.txt | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index c1ac70cb0afb..c30934dd975b 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -5,11 +5,6 @@ Required properties: - compatible - "samsung,odroidxu3-audio" - for Odroid XU3 board, "samsung,odroidxu4-audio" - for Odroid XU4 board - model - the user-visible name of this sound complex - - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S - controller - - 'codec' subnode with a 'sound-dai' property containing list of phandles - to the CODEC nodes, first entry must be corresponding to the MAX98090 - CODEC and the second entry must be the phandle of the HDMI IP block node - clocks - should contain entries matching clock names in the clock-names property - clock-names - should contain following entries: @@ -32,12 +27,18 @@ Required properties: For Odroid XU4: no entries +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be corresponding to the MAX98090 + CODEC and the second entry must be the phandle of the HDMI IP block node + Example: sound { compatible = "samsung,odroidxu3-audio"; - samsung,cpu-dai = <&i2s0>; - samsung,codec-dai = <&max98090>; model = "Odroid-XU3"; samsung,audio-routing = "Headphone Jack", "HPL", -- cgit v1.2.3 From fc05a5b222530617d99d0e803abb262130fdb0c4 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Tue, 13 Jun 2017 15:27:46 +0800 Subject: ASoC: rockchip: add support for pdm controller The Pulse Density Modulation Interface Controller (PDMC) is a PDM interface controller and decoder that support PDM format. It integrates a clock generator driving the PDM microphone and embeds filters which decimate the incoming bit stream to obtain most common audio rates. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip,pdm.txt | 39 ++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_pdm.c | 516 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.h | 83 ++++ 5 files changed, 649 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 sound/soc/rockchip/rockchip_pdm.c create mode 100644 sound/soc/rockchip/rockchip_pdm.h (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.txt b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt new file mode 100644 index 000000000000..921729de7346 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt @@ -0,0 +1,39 @@ +* Rockchip PDM controller + +Required properties: + +- compatible: "rockchip,pdm" +- reg: physical base address of the controller and length of memory mapped + region. +- dmas: DMA specifiers for rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain following: + - "pdm_hclk": clock for PDM BUS + - "pdm_clk" : clock for PDM controller +- pinctrl-names: Must contain a "default" entry. +- pinctrl-N: One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. + +Example for rk3328 PDM controller: + +pdm: pdm@ff040000 { + compatible = "rockchip,pdm"; + reg = <0x0 0xff040000 0x0 0x1000>; + clocks = <&clk_pdm>, <&clk_gates28 0>; + clock-names = "pdm_clk", "pdm_hclk"; + dmas = <&pdma 16>; + #dma-cells = <1>; + dma-names = "rx"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pdmm0_clk + &pdmm0_fsync + &pdmm0_sdi0 + &pdmm0_sdi1 + &pdmm0_sdi2 + &pdmm0_sdi3>; + pinctrl-1 = <&pdmm0_sleep>; + status = "disabled"; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index e3ca1e973de5..c84487805876 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -15,6 +15,15 @@ config SND_SOC_ROCKCHIP_I2S Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. +config SND_SOC_ROCKCHIP_PDM + tristate "Rockchip PDM Controller Driver" + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for PDM driver for + Rockchip PDM Controller. The Controller supports up to maximum of + 8 channels record. + config SND_SOC_ROCKCHIP_SPDIF tristate "Rockchip SPDIF Device Driver" depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 991f91bea9f9..105f0e14a4ab 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,8 +1,10 @@ # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c new file mode 100644 index 000000000000..c5ddeed97260 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -0,0 +1,516 @@ +/* + * Rockchip PDM ALSA SoC Digital Audio Interface(DAI) driver + * + * Copyright (C) 2017 Fuzhou Rockchip Electronics Co., Ltd + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "rockchip_pdm.h" + +#define PDM_DMA_BURST_SIZE (16) /* size * width: 16*4 = 64 bytes */ + +struct rk_pdm_dev { + struct device *dev; + struct clk *clk; + struct clk *hclk; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data capture_dma_data; +}; + +struct rk_pdm_clkref { + unsigned int sr; + unsigned int clk; +}; + +static struct rk_pdm_clkref clkref[] = { + { 8000, 40960000 }, + { 11025, 56448000 }, + { 12000, 61440000 }, +}; + +static unsigned int get_pdm_clk(unsigned int sr) +{ + unsigned int i, count, clk, div; + + clk = 0; + if (!sr) + return clk; + + count = ARRAY_SIZE(clkref); + for (i = 0; i < count; i++) { + if (sr % clkref[i].sr) + continue; + div = sr / clkref[i].sr; + if ((div & (div - 1)) == 0) { + clk = clkref[i].clk; + break; + } + } + + return clk; +} + +static inline struct rk_pdm_dev *to_info(struct snd_soc_dai *dai) +{ + return snd_soc_dai_get_drvdata(dai); +} + +static void rockchip_pdm_rxctrl(struct rk_pdm_dev *pdm, int on) +{ + if (on) { + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, + PDM_DMA_RD_MSK, PDM_DMA_RD_EN); + regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, + PDM_RX_MASK, PDM_RX_START); + } else { + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, + PDM_DMA_RD_MSK, PDM_DMA_RD_DIS); + regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, + PDM_RX_MASK | PDM_RX_CLR_MASK, + PDM_RX_STOP | PDM_RX_CLR_WR); + } +} + +static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk_pdm_dev *pdm = to_info(dai); + unsigned int val = 0; + unsigned int clk_rate, clk_div, samplerate; + int ret; + + samplerate = params_rate(params); + clk_rate = get_pdm_clk(samplerate); + if (!clk_rate) + return -EINVAL; + + ret = clk_set_rate(pdm->clk, clk_rate); + if (ret) + return -EINVAL; + + clk_div = DIV_ROUND_CLOSEST(clk_rate, samplerate); + + switch (clk_div) { + case 320: + val = PDM_CLK_320FS; + break; + case 640: + val = PDM_CLK_640FS; + break; + case 1280: + val = PDM_CLK_1280FS; + break; + case 2560: + val = PDM_CLK_2560FS; + break; + case 5120: + val = PDM_CLK_5120FS; + break; + default: + dev_err(pdm->dev, "unsupported div: %d\n", clk_div); + return -EINVAL; + } + + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_DS_RATIO_MSK, val); + regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, + PDM_HPF_CF_MSK, PDM_HPF_60HZ); + regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, + PDM_HPF_LE | PDM_HPF_RE, PDM_HPF_LE | PDM_HPF_RE); + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_CLK_EN, PDM_CLK_EN); + + val = 0; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + val |= PDM_VDW(8); + break; + case SNDRV_PCM_FORMAT_S16_LE: + val |= PDM_VDW(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= PDM_VDW(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= PDM_VDW(24); + break; + case SNDRV_PCM_FORMAT_S32_LE: + val |= PDM_VDW(32); + break; + default: + return -EINVAL; + } + + switch (params_channels(params)) { + case 8: + val |= PDM_PATH3_EN; + /* fallthrough */ + case 6: + val |= PDM_PATH2_EN; + /* fallthrough */ + case 4: + val |= PDM_PATH1_EN; + /* fallthrough */ + case 2: + val |= PDM_PATH0_EN; + break; + default: + dev_err(pdm->dev, "invalid channel: %d\n", + params_channels(params)); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + regmap_update_bits(pdm->regmap, PDM_CTRL0, + PDM_PATH_MSK | PDM_VDW_MSK, + val); + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, PDM_DMA_RDL_MSK, + PDM_DMA_RDL(16)); + regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, + PDM_RX_MASK | PDM_RX_CLR_MASK, + PDM_RX_STOP | PDM_RX_CLR_WR); + } + + return 0; +} + +static int rockchip_pdm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct rk_pdm_dev *pdm = to_info(cpu_dai); + unsigned int mask = 0, val = 0; + + mask = PDM_CKP_MSK; + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val = PDM_CKP_NORMAL; + break; + case SND_SOC_DAIFMT_IB_NF: + val = PDM_CKP_INVERTED; + break; + default: + return -EINVAL; + } + + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, mask, val); + + return 0; +} + +static int rockchip_pdm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct rk_pdm_dev *pdm = to_info(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_pdm_rxctrl(pdm, 1); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_pdm_rxctrl(pdm, 0); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rockchip_pdm_dai_probe(struct snd_soc_dai *dai) +{ + struct rk_pdm_dev *pdm = to_info(dai); + + dai->capture_dma_data = &pdm->capture_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops rockchip_pdm_dai_ops = { + .set_fmt = rockchip_pdm_set_fmt, + .trigger = rockchip_pdm_trigger, + .hw_params = rockchip_pdm_hw_params, +}; + +#define ROCKCHIP_PDM_RATES SNDRV_PCM_RATE_8000_192000 +#define ROCKCHIP_PDM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rockchip_pdm_dai = { + .probe = rockchip_pdm_dai_probe, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = ROCKCHIP_PDM_RATES, + .formats = ROCKCHIP_PDM_FORMATS, + }, + .ops = &rockchip_pdm_dai_ops, + .symmetric_rates = 1, +}; + +static const struct snd_soc_component_driver rockchip_pdm_component = { + .name = "rockchip-pdm", +}; + +static int rockchip_pdm_runtime_suspend(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + + clk_disable_unprepare(pdm->clk); + clk_disable_unprepare(pdm->hclk); + + return 0; +} + +static int rockchip_pdm_runtime_resume(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(pdm->clk); + if (ret) { + dev_err(pdm->dev, "clock enable failed %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(pdm->hclk); + if (ret) { + dev_err(pdm->dev, "hclock enable failed %d\n", ret); + return ret; + } + + return 0; +} + +static bool rockchip_pdm_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_SYSCONFIG: + case PDM_CTRL0: + case PDM_CTRL1: + case PDM_CLK_CTRL: + case PDM_HPF_CTRL: + case PDM_FIFO_CTRL: + case PDM_DMA_CTRL: + case PDM_INT_EN: + case PDM_INT_CLR: + case PDM_DATA_VALID: + return true; + default: + return false; + } +} + +static bool rockchip_pdm_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_SYSCONFIG: + case PDM_CTRL0: + case PDM_CTRL1: + case PDM_CLK_CTRL: + case PDM_HPF_CTRL: + case PDM_FIFO_CTRL: + case PDM_DMA_CTRL: + case PDM_INT_EN: + case PDM_INT_CLR: + case PDM_INT_ST: + case PDM_DATA_VALID: + case PDM_VERSION: + return true; + default: + return false; + } +} + +static bool rockchip_pdm_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_SYSCONFIG: + case PDM_INT_CLR: + case PDM_INT_ST: + return true; + default: + return false; + } +} + +static const struct regmap_config rockchip_pdm_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = PDM_VERSION, + .writeable_reg = rockchip_pdm_wr_reg, + .readable_reg = rockchip_pdm_rd_reg, + .volatile_reg = rockchip_pdm_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int rockchip_pdm_probe(struct platform_device *pdev) +{ + struct rk_pdm_dev *pdm; + struct resource *res; + void __iomem *regs; + int ret; + + pdm = devm_kzalloc(&pdev->dev, sizeof(*pdm), GFP_KERNEL); + if (!pdm) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + pdm->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &rockchip_pdm_regmap_config); + if (IS_ERR(pdm->regmap)) + return PTR_ERR(pdm->regmap); + + pdm->capture_dma_data.addr = res->start + PDM_RXFIFO_DATA; + pdm->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + pdm->capture_dma_data.maxburst = PDM_DMA_BURST_SIZE; + + pdm->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, pdm); + + pdm->clk = devm_clk_get(&pdev->dev, "pdm_clk"); + if (IS_ERR(pdm->clk)) + return PTR_ERR(pdm->clk); + + pdm->hclk = devm_clk_get(&pdev->dev, "pdm_hclk"); + if (IS_ERR(pdm->hclk)) + return PTR_ERR(pdm->hclk); + + ret = clk_prepare_enable(pdm->hclk); + if (ret) + return ret; + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = rockchip_pdm_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &rockchip_pdm_component, + &rockchip_pdm_dai, 1); + + if (ret) { + dev_err(&pdev->dev, "could not register dai: %d\n", ret); + goto err_suspend; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "could not register pcm: %d\n", ret); + goto err_suspend; + } + + return 0; + +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + rockchip_pdm_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); + + clk_disable_unprepare(pdm->hclk); + + return ret; +} + +static int rockchip_pdm_remove(struct platform_device *pdev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + rockchip_pdm_runtime_suspend(&pdev->dev); + + clk_disable_unprepare(pdm->clk); + clk_disable_unprepare(pdm->hclk); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int rockchip_pdm_suspend(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + + regcache_mark_dirty(pdm->regmap); + + return 0; +} + +static int rockchip_pdm_resume(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + + ret = regcache_sync(pdm->regmap); + + pm_runtime_put(dev); + + return ret; +} +#endif + +static const struct dev_pm_ops rockchip_pdm_pm_ops = { + SET_RUNTIME_PM_OPS(rockchip_pdm_runtime_suspend, + rockchip_pdm_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(rockchip_pdm_suspend, rockchip_pdm_resume) +}; + +static const struct of_device_id rockchip_pdm_match[] = { + { .compatible = "rockchip,pdm", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rockchip_pdm_match); + +static struct platform_driver rockchip_pdm_driver = { + .probe = rockchip_pdm_probe, + .remove = rockchip_pdm_remove, + .driver = { + .name = "rockchip-pdm", + .of_match_table = of_match_ptr(rockchip_pdm_match), + .pm = &rockchip_pdm_pm_ops, + }, +}; + +module_platform_driver(rockchip_pdm_driver); + +MODULE_AUTHOR("Sugar "); +MODULE_DESCRIPTION("Rockchip PDM Controller Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pdm.h b/sound/soc/rockchip/rockchip_pdm.h new file mode 100644 index 000000000000..886b48d128fd --- /dev/null +++ b/sound/soc/rockchip/rockchip_pdm.h @@ -0,0 +1,83 @@ +/* + * Rockchip PDM ALSA SoC Digital Audio Interface(DAI) driver + * + * Copyright (C) 2017 Fuzhou Rockchip Electronics Co., Ltd + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef _ROCKCHIP_PDM_H +#define _ROCKCHIP_PDM_H + +/* PDM REGS */ +#define PDM_SYSCONFIG (0x0000) +#define PDM_CTRL0 (0x0004) +#define PDM_CTRL1 (0x0008) +#define PDM_CLK_CTRL (0x000c) +#define PDM_HPF_CTRL (0x0010) +#define PDM_FIFO_CTRL (0x0014) +#define PDM_DMA_CTRL (0x0018) +#define PDM_INT_EN (0x001c) +#define PDM_INT_CLR (0x0020) +#define PDM_INT_ST (0x0024) +#define PDM_RXFIFO_DATA (0x0030) +#define PDM_DATA_VALID (0x0054) +#define PDM_VERSION (0x0058) + +/* PDM_SYSCONFIG */ +#define PDM_RX_MASK (0x1 << 2) +#define PDM_RX_START (0x1 << 2) +#define PDM_RX_STOP (0x0 << 2) +#define PDM_RX_CLR_MASK (0x1 << 0) +#define PDM_RX_CLR_WR (0x1 << 0) +#define PDM_RX_CLR_DONE (0x0 << 0) + +/* PDM CTRL0 */ +#define PDM_PATH_MSK (0xf << 27) +#define PDM_PATH3_EN BIT(30) +#define PDM_PATH2_EN BIT(29) +#define PDM_PATH1_EN BIT(28) +#define PDM_PATH0_EN BIT(27) +#define PDM_HWT_EN BIT(26) +#define PDM_VDW_MSK (0x1f << 0) +#define PDM_VDW(X) ((X - 1) << 0) + +/* PDM CLK CTRL */ +#define PDM_CLK_MSK BIT(5) +#define PDM_CLK_EN BIT(5) +#define PDM_CLK_DIS (0x0 << 5) +#define PDM_CKP_MSK BIT(3) +#define PDM_CKP_NORMAL (0x0 << 3) +#define PDM_CKP_INVERTED BIT(3) +#define PDM_DS_RATIO_MSK (0x7 << 0) +#define PDM_CLK_320FS (0x0 << 0) +#define PDM_CLK_640FS (0x1 << 0) +#define PDM_CLK_1280FS (0x2 << 0) +#define PDM_CLK_2560FS (0x3 << 0) +#define PDM_CLK_5120FS (0x4 << 0) + +/* PDM HPF CTRL */ +#define PDM_HPF_LE BIT(3) +#define PDM_HPF_RE BIT(2) +#define PDM_HPF_CF_MSK (0x3 << 0) +#define PDM_HPF_3P79HZ (0x0 << 0) +#define PDM_HPF_60HZ (0x1 << 0) +#define PDM_HPF_243HZ (0x2 << 0) +#define PDM_HPF_493HZ (0x3 << 0) + +/* PDM DMA CTRL */ +#define PDM_DMA_RD_MSK BIT(8) +#define PDM_DMA_RD_EN BIT(8) +#define PDM_DMA_RD_DIS (0x0 << 8) +#define PDM_DMA_RDL_MSK (0x7f << 0) +#define PDM_DMA_RDL(X) ((X - 1) << 0) + +#endif /* _ROCKCHIP_PDM_H */ -- cgit v1.2.3 From 55f42d2e28a42b06907c916c3c71ceb6dfb5afc4 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Fri, 9 Jun 2017 15:59:32 +0800 Subject: ASoC: rockchip: add bindings for spdif controller this patch add compatible for rk3228/rk3328 spdif, Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-spdif.txt | 2 ++ sound/soc/rockchip/rockchip_spdif.c | 4 ++++ 2 files changed, 6 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt index 11046429a118..4706b96d450b 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -9,7 +9,9 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-spdif" - "rockchip,rk3188-spdif" + - "rockchip,rk3228-spdif" - "rockchip,rk3288-spdif" + - "rockchip,rk3328-spdif" - "rockchip,rk3366-spdif" - "rockchip,rk3368-spdif" - "rockchip,rk3399-spdif" diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index fa8101d1e16f..ee5055d47d13 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -49,8 +49,12 @@ static const struct of_device_id rk_spdif_match[] = { .data = (void *)RK_SPDIF_RK3066 }, { .compatible = "rockchip,rk3188-spdif", .data = (void *)RK_SPDIF_RK3188 }, + { .compatible = "rockchip,rk3228-spdif", + .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3288-spdif", .data = (void *)RK_SPDIF_RK3288 }, + { .compatible = "rockchip,rk3328-spdif", + .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3366-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3368-spdif", -- cgit v1.2.3 From 5b16c8b1faf4bf77934c0a206cfbe77154c79fd7 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:15:28 +0200 Subject: ASoC: stm32: sai: fix DT example Correct the device tree example. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,stm32-sai.txt | 25 +++++++++------------- 1 file changed, 10 insertions(+), 15 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index c59a3d779e06..a0feeed1710e 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -36,6 +36,10 @@ SAI subnodes required properties: - pinctrl-names: should contain only value "default" - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt +The device node should contain one 'port' child node with one child 'endpoint' +node, according to the bindings defined in Documentation/devicetree/bindings/ +graph.txt. + Example: sound_card { compatible = "audio-graph-card"; @@ -46,16 +50,15 @@ sai1: sai1@40015800 { compatible = "st,stm32f4-sai"; #address-cells = <1>; #size-cells = <1>; - ranges; + ranges = <0 0x40015800 0x400>; reg = <0x40015800 0x4>; clocks = <&rcc 1 CLK_SAIQ_PDIV>, <&rcc 1 CLK_I2SQ_PDIV>; clock-names = "x8k", "x11k"; interrupts = <87>; sai1b: audio-controller@40015824 { - #sound-dai-cells = <0>; compatible = "st,stm32-sai-sub-b"; - reg = <0x40015824 0x1C>; + reg = <0x24 0x1C>; clocks = <&rcc 1 CLK_SAI2>; clock-names = "sai_ck"; dmas = <&dma2 5 0 0x400 0x0>; @@ -63,18 +66,10 @@ sai1: sai1@40015800 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai1b>; - ports { - #address-cells = <1>; - #size-cells = <0>; - - sai1b_port: port@0 { - reg = <0>; - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - audio-graph-card,format = "i2s"; - audio-graph-card,bitclock-master = <&codec_endpoint>; - audio-graph-card,frame-master = <&codec_endpoint>; - }; + sai1b_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + format = "i2s"; }; }; }; -- cgit v1.2.3 From 3861da5801f59f3e9252b6a5db92cfa71629995c Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:16:23 +0200 Subject: ASoC: stm32: add h7 support for sai Document device tree bindings for STM32H7 SAI. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,stm32-sai.txt | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index a0feeed1710e..f1c5ae59e7c9 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -6,7 +6,7 @@ The SAI contains two independent audio sub-blocks. Each sub-block has its own clock generator and I/O lines controller. Required properties: - - compatible: Should be "st,stm32f4-sai" + - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - reg: Base address and size of SAI common register set. - clocks: Must contain phandle and clock specifier pairs for each entry in clock-names. @@ -47,24 +47,24 @@ sound_card { }; sai1: sai1@40015800 { - compatible = "st,stm32f4-sai"; + compatible = "st,stm32h7-sai"; #address-cells = <1>; #size-cells = <1>; ranges = <0 0x40015800 0x400>; reg = <0x40015800 0x4>; - clocks = <&rcc 1 CLK_SAIQ_PDIV>, <&rcc 1 CLK_I2SQ_PDIV>; + clocks = <&rcc PLL1_Q>, <&rcc PLL2_P>; clock-names = "x8k", "x11k"; interrupts = <87>; - sai1b: audio-controller@40015824 { - compatible = "st,stm32-sai-sub-b"; - reg = <0x24 0x1C>; - clocks = <&rcc 1 CLK_SAI2>; + sai1a: audio-controller@40015804 { + compatible = "st,stm32-sai-sub-a"; + reg = <0x4 0x1C>; + clocks = <&rcc SAI1_CK>; clock-names = "sai_ck"; - dmas = <&dma2 5 0 0x400 0x0>; + dmas = <&dmamux1 1 87 0x400 0x0>; dma-names = "tx"; pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1b>; + pinctrl-0 = <&pinctrl_sai1a>; sai1b_port: port { cpu_endpoint: endpoint { -- cgit v1.2.3 From 0507cb0226acfd7ba114c59f6a76fdc7a1c6b01e Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Tue, 20 Jun 2017 11:58:46 +0200 Subject: ASoC: stm32: Add DT bindings for SPDIFRX interface Add documentation of device tree bindings for the STM32 SPDIFRX interface. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 ++++++++++++++++++++++ 1 file changed, 56 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt new file mode 100644 index 000000000000..33826f2459fa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt @@ -0,0 +1,56 @@ +STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). + +The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with +IEC-60958 and IEC-61937. + +Required properties: + - compatible: should be "st,stm32h7-spdifrx" + - reg: cpu DAI IP base address and size + - clocks: must contain an entry for kclk (used as S/PDIF signal reference) + - clock-names: must contain "kclk" + - interrupts: cpu DAI interrupt line + - dmas: DMA specifiers for audio data DMA and iec control flow DMA + See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt + - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" + +Optional properties: + - resets: Reference to a reset controller asserting the SPDIFRX + +The device node should contain one 'port' child node with one child 'endpoint' +node, according to the bindings defined in Documentation/devicetree/bindings/ +graph.txt. + +Example: +spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIFRX_CK>; + clock-names = "kclk"; + interrupts = <97>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + + spdifrx_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + }; + }; +}; + +spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + + codec_port: port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; +}; + +soundcard { + compatible = "audio-graph-card"; + dais = <&spdifrx_port>; +}; -- cgit v1.2.3 From 7de35c122e2dd8dc4d74b3782ced9c03115dc268 Mon Sep 17 00:00:00 2001 From: Baoyou Xie Date: Thu, 22 Jun 2017 14:51:57 +0800 Subject: ASoC: add bindings for ZTE zx-aud96p22 audio codec It adds dt-bindings document for ZTE zx-aud96p22 audio codec. Signed-off-by: Baoyou Xie Acked-by: Rob Herring Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/zte,zx-aud96p22.txt | 24 ++++++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt b/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt new file mode 100644 index 000000000000..41bb1040eb71 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt @@ -0,0 +1,24 @@ +ZTE ZX AUD96P22 Audio Codec + +Required properties: + - compatible: Must be "zte,zx-aud96p22" + - #sound-dai-cells: Should be 0 + - reg: I2C bus slave address of AUD96P22 + +Example: + + i2c0: i2c@1486000 { + compatible = "zte,zx296718-i2c"; + reg = <0x01486000 0x1000>; + interrupts = ; + #address-cells = <1>; + #size-cells = <0>; + clocks = <&audiocrm AUDIO_I2C0_WCLK>; + clock-frequency = <1600000>; + + aud96p22: codec@22 { + compatible = "zte,zx-aud96p22"; + #sound-dai-cells = <0>; + reg = <0x22>; + }; + }; -- cgit v1.2.3 From f1f940490d3ccff96da9cc81d57c2c083c398a18 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 22 Jun 2017 06:22:49 +0000 Subject: ASoC: audio-graph-scu-card: support 2nd codec endpoint on DT audio-graph-scu-card can handle below connection which is mainly for sound mixing purpose. +----------+ +-------+ | CPU0--+--|-->| Codec | | | | +-------+ | CPU1--+ | +----------+ >From OF-graph point of view, it should have CPU0 <-> Codec, and CPU1 <-> Codec on DT. But current driver doesn't care about 2nd connection of Codec, because it is dummy from DPCM point of view. This patch can care 2nd Codec connection, and it should be supported from OF-graph point of view. It still have backward compatibility. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/audio-graph-scu-card.txt | 9 +++++-- sound/soc/generic/audio-graph-scu-card.c | 28 +++++++++++++++++++--- 2 files changed, 32 insertions(+), 5 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b63c5594bbb3..8b8afe9fcb31 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -90,9 +90,12 @@ Example 2. 2 CPU 1 Codec (Mixing) ... port { - codec_endpoint: endpoint { + codec_endpoint0: endpoint { remote-endpoint = <&cpu_endpoint0>; }; + codec_endpoint1: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; }; }; @@ -101,7 +104,7 @@ Example 2. 2 CPU 1 Codec (Mixing) ports { cpu_port0: port { cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint>; + remote-endpoint = <&codec_endpoint0>; dai-format = "left_j"; ... @@ -109,6 +112,8 @@ Example 2. 2 CPU 1 Codec (Mixing) }; cpu_port1: port { cpu_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint1>; + dai-format = "left_j"; ... }; diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 061c7a60d6b4..dcd2df37bc3b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -183,6 +183,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *cpu_ep; struct device_node *codec_ep; struct device_node *rcpu_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; unsigned int daifmt = 0; int dai_idx, ret; int rc, codec; @@ -235,6 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } dai_idx = 0; + codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* * To listup valid sounds continuously, @@ -245,15 +248,22 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (codec) { - if (!codec_ep) + if (!codec_port) continue; + if (codec_port_old == codec_port) + continue; + + codec_port_old = codec_port; + /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); if (ret < 0) @@ -284,22 +294,34 @@ static int asoc_graph_get_dais_count(struct device *dev) struct device_node *cpu_port; struct device_node *cpu_ep; struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; int count = 0; int rc; + codec_port_old = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (cpu_ep) count++; - if (codec_ep) - count++; + + if (!codec_port) + continue; + + if (codec_port_old == codec_port) + continue; + + count++; + codec_port_old = codec_port; } return count; -- cgit v1.2.3 From e3839bd6f56a291f00a4c3737eb15ca0344a82a9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 19 Jun 2017 00:39:29 +0000 Subject: drm: dw-hdmi-i2s: add .get_dai_id callback for ALSA SoC ALSA SoC needs to know connected DAI ID for probing. It is not a big problem if device/driver was only for sound, but getting DAI ID will be difficult if device includes both Video/Sound, like HDMI. To solve this issue, this patch adds new .get_dai_id callback on hdmi_codec_ops. dw-hdmi-i2s will assume that HDMI sound will be connected to reg = <2>. Then, ALSA SoC side will recognized it as DAI 0 ports { #address-cells = <1>; #size-cells = <0>; port@0 { reg = <0>; /* HDMI Video IN */ }; port@1 { reg = <1>; /* HDMI OUT */ }; port@2 { reg = <2>; /* HDMI Sound IN */ }; }; Signed-off-by: Kuninori Morimoto Acked-by: Archit Taneja Signed-off-by: Mark Brown --- .../bindings/display/bridge/renesas,dw-hdmi.txt | 9 ++++++++- drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 21 +++++++++++++++++++++ 2 files changed, 29 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt b/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt index f6b3f36d422b..81b68580e199 100644 --- a/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt +++ b/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt @@ -25,7 +25,8 @@ Required properties: - clock-names: Shall contain "iahb" and "isfr" as defined in dw_hdmi.txt. - ports: See dw_hdmi.txt. The DWC HDMI shall have one port numbered 0 corresponding to the video input of the controller and one port numbered 1 - corresponding to its HDMI output. Each port shall have a single endpoint. + corresponding to its HDMI output, and one port numbered 2 corresponding to + sound input of the controller. Each port shall have a single endpoint. Optional properties: @@ -59,6 +60,12 @@ Example: remote-endpoint = <&hdmi0_con>; }; }; + port@2 { + reg = <2>; + rcar_dw_hdmi0_sound_in: endpoint { + remote-endpoint = <&hdmi_sound_out>; + }; + }; }; }; diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index aaf287d2e91d..b2cf59f54c88 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -82,9 +82,30 @@ static void dw_hdmi_i2s_audio_shutdown(struct device *dev, void *data) hdmi_write(audio, HDMI_AUD_CONF0_SW_RESET, HDMI_AUD_CONF0); } +static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + struct of_endpoint of_ep; + int ret; + + ret = of_graph_parse_endpoint(endpoint, &of_ep); + if (ret < 0) + return ret; + + /* + * HDMI sound should be located as reg = <2> + * Then, it is sound port 0 + */ + if (of_ep.port == 2) + return 0; + + return -EINVAL; +} + static struct hdmi_codec_ops dw_hdmi_i2s_ops = { .hw_params = dw_hdmi_i2s_hw_params, .audio_shutdown = dw_hdmi_i2s_audio_shutdown, + .get_dai_id = dw_hdmi_i2s_get_dai_id, }; static int snd_dw_hdmi_probe(struct platform_device *pdev) -- cgit v1.2.3 From 7204e97685634813d8456f1900b7f38fa7701e60 Mon Sep 17 00:00:00 2001 From: John Stultz Date: Tue, 13 Jun 2017 14:59:49 -0700 Subject: drm: adv7511_audio: Add .get_dai_id callback to map port number to dai id ALSA SoC needs to know connected DAI ID for probing. Using the new audio-card-graph approach, ports/endpoints are used to describe how the links are connected. Unfortunately, since ports/endpoints are used as well for video linkages, there are some issues mixing the port ids to the two (video and audio) namespaces. To solve this issue, this patch adds new .get_dai_id callback on hdmi_codec_ops. The will assume that HDMI audio out will be connected to reg = <2>. This will then be remapped to the ALSA SoC side will as DAI 0. Allowing the adv7511's hdmi audio support to be used with the audio-card-graph. Credit to Kuninori Morimoto who's patch to dw-hdmi-i2s-audio.c was what this was mostly copy-pasted from. Cc: Kuninori Morimoto Cc: Archit Taneja Cc: Mark Brown Cc: Rob Herring Cc: David Airlie Cc: Lars-Peter Clausen Cc: Linux-ALSA Cc: dri-devel@lists.freedesktop.org Signed-off-by: John Stultz Signed-off-by: Mark Brown --- .../bindings/display/bridge/adi,adv7511.txt | 8 ++++++++ drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 22 ++++++++++++++++++++++ 2 files changed, 30 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt index 00ea670b8c4d..06668bca7ffc 100644 --- a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt +++ b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt @@ -78,6 +78,7 @@ graph bindings specified in Documentation/devicetree/bindings/graph.txt. remote endpoint phandle should be a reference to a valid mipi_dsi_host device node. - Video port 1 for the HDMI output +- Audio port 2 for the HDMI audio input Example @@ -112,5 +113,12 @@ Example remote-endpoint = <&hdmi_connector_in>; }; }; + + port@2 { + reg = <2>; + codec_endpoint: endpoint { + remote-endpoint = <&i2s0_cpu_endpoint>; + }; + }; }; }; diff --git a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c index cf92ebfe6ab7..67469c26bae8 100644 --- a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c +++ b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c @@ -11,6 +11,7 @@ #include #include #include +#include #include "adv7511.h" @@ -182,10 +183,31 @@ static void audio_shutdown(struct device *dev, void *data) { } +static int adv7511_hdmi_i2s_get_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + struct of_endpoint of_ep; + int ret; + + ret = of_graph_parse_endpoint(endpoint, &of_ep); + if (ret < 0) + return ret; + + /* + * HDMI sound should be located as reg = <2> + * Then, it is sound port 0 + */ + if (of_ep.port == 2) + return 0; + + return -EINVAL; +} + static const struct hdmi_codec_ops adv7511_codec_ops = { .hw_params = adv7511_hdmi_hw_params, .audio_shutdown = audio_shutdown, .audio_startup = audio_startup, + .get_dai_id = adv7511_hdmi_i2s_get_dai_id, }; static struct hdmi_codec_pdata codec_data = { -- cgit v1.2.3 From 8a70b4544ef4f094cc2c52734e097cc358f56603 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 29 Jun 2017 14:22:24 +0100 Subject: ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs. Add some DAPM widget types to better support the construction of DAPM graphs within DSPs. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 18 ++++++++++++++++++ include/sound/soc-dapm.h | 7 +++++++ include/uapi/sound/asoc.h | 10 +++++++++- sound/soc/soc-topology.c | 8 ++++++++ 4 files changed, 42 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index a27f42befa4d..8e44107933ab 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -105,6 +105,24 @@ Pre Special PRE widget (exec before all others) Post Special POST widget (exec after all others) +Buffer + Inter widget audio data buffer within a DSP. +Scheduler + DSP internal scheduler that schedules component/pipeline processing + work. +Effect + Widget that performs an audio processing effect. +SRC + Sample Rate Converter within DSP or CODEC +ASRC + Asynchronous Sample Rate Converter within DSP or CODEC +Encoder + Widget that encodes audio data from one format (usually PCM) to another + usually more compressed format. +Decoder + Widget that decodes audio data from a compressed format to an + uncompressed format like PCM. + (Widgets are defined in include/sound/soc-dapm.h) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..344b96c206a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,13 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_buffer, /* DSP/CODEC internal buffer */ + snd_soc_dapm_scheduler, /* DSP/CODEC internal scheduler */ + snd_soc_dapm_effect, /* DSP/CODEC effect component */ + snd_soc_dapm_src, /* DSP/CODEC SRC component */ + snd_soc_dapm_asrc, /* DSP/CODEC ASRC component */ + snd_soc_dapm_encoder, /* FW/SW audio encoder component */ + snd_soc_dapm_decoder, /* FW/SW audio decoder component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..78014ec56357 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,15 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_SCHEDULER 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_ENCODER 22 +#define SND_SOC_TPLG_DAPM_DECODER 23 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DECODER /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 002772e3ba2c..dd3a391476ae 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,14 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_BUFFER, snd_soc_dapm_buffer}, + {SND_SOC_TPLG_DAPM_SCHEDULER, snd_soc_dapm_scheduler}, + {SND_SOC_TPLG_DAPM_EFFECT, snd_soc_dapm_effect}, + {SND_SOC_TPLG_DAPM_SIGGEN, snd_soc_dapm_siggen}, + {SND_SOC_TPLG_DAPM_SRC, snd_soc_dapm_src}, + {SND_SOC_TPLG_DAPM_ASRC, snd_soc_dapm_asrc}, + {SND_SOC_TPLG_DAPM_ENCODER, snd_soc_dapm_encoder}, + {SND_SOC_TPLG_DAPM_DECODER, snd_soc_dapm_decoder}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- cgit v1.2.3 From fc3ba81a5adac413312019413c91b1e6a5d8d1fa Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 29 Jun 2017 11:41:30 +0800 Subject: ASoC: nau8825: change crosstalk-bypass property to bool type The property type of "nuvoton,crosstalk-bypass" changes to boolean. The document is updated as well. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/nau8825.txt | 3 +++ sound/soc/codecs/nau8825.c | 6 ++---- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index d3374231c871..2f5e973285a6 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,6 +69,8 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,crosstalk-bypass: make crosstalk function bypass if set. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock @@ -96,6 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-bypass; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 80bae481e75d..46a30eaa7ace 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2506,10 +2506,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", - &nau8825->xtalk_bypass); - if (ret) - nau8825->xtalk_bypass = 1; + nau8825->xtalk_bypass = device_property_read_bool(dev, + "nuvoton,crosstalk-bypass"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- cgit v1.2.3 From a0c683d734e0b3589892c17d0e1187f20d2c3a54 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 29 Jun 2017 21:26:37 +0800 Subject: ASoC: audio-graph-card: update bindings for amplifier support The audio-graph-card should be able to support widgets and routing in the same way as what simple-audio-card does. The patch adds the properties into audio-graph-card bindings. Then an optional property 'pa-gpios' for controlling external amplifier, which depends on DAPM widgets and routing, is added. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/audio-graph-card.txt | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt index bac4b1b1060f..6e6720aa33f1 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -10,6 +10,8 @@ see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt Below are same as Simple-Card. - label +- widgets +- routing - dai-format - frame-master - bitclock-master @@ -24,6 +26,9 @@ Required properties: - compatible : "audio-graph-card"; - dais : list of CPU DAI port{s} +Optional properties: +- pa-gpios: GPIO used to control external amplifier. + Example: Single DAI case sound_card { -- cgit v1.2.3 From d8b53bff0a499cd05a5026307af9a5f41f604ea3 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 2 Jul 2017 11:44:43 +0900 Subject: ALSA: pcm: add a documentation for tracepoints In PCM interface/protocol for userspace, parameters of runtime for PCM substream is decided by an interaction between applications and ALSA PCM core. In former commits, some tracepoints were added to probe a part of the interaction. This commit adds a documentation about the interaction and the tracepoints. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/tracepoints.rst | 172 ++++++++++++++++++++++++++++ 2 files changed, 173 insertions(+) create mode 100644 Documentation/sound/designs/tracepoints.rst (limited to 'Documentation') diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 04dcdae3e4f2..f0749943ccb2 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -9,6 +9,7 @@ Designs and Implementations compress-offload timestamping jack-controls + tracepoints procfile powersave oss-emulation diff --git a/Documentation/sound/designs/tracepoints.rst b/Documentation/sound/designs/tracepoints.rst new file mode 100644 index 000000000000..78bc5572f829 --- /dev/null +++ b/Documentation/sound/designs/tracepoints.rst @@ -0,0 +1,172 @@ +=================== +Tracepoints in ALSA +=================== + +2017/07/02 +Takasahi Sakamoto + +Tracepoints in ALSA PCM core +============================ + +ALSA PCM core registers ``snd_pcm`` subsystem to kernel tracepoint system. +This subsystem includes two categories of tracepoints; for state of PCM buffer +and for processing of PCM hardware parameters. These tracepoints are available +when corresponding kernel configurations are enabled. When ``CONFIG_SND_DEBUG`` +is enabled, the latter tracepoints are available. When additional +``SND_PCM_XRUN_DEBUG`` is enabled too, the former trace points are enabled. + +Tracepoints for state of PCM buffer +------------------------------------ + +This category includes four tracepoints; ``hwptr``, ``applptr``, ``xrun`` and +``hw_ptr_error``. + +Tracepoints for processing of PCM hardware parameters +----------------------------------------------------- + +This category includes two tracepoints; ``hw_mask_param`` and +``hw_interval_param``. + +In a design of ALSA PCM core, data transmission is abstracted as PCM substream. +Applications manage PCM substream to maintain data transmission for PCM frames. +Before starting the data transmission, applications need to configure PCM +substream. In this procedure, PCM hardware parameters are decided by +interaction between applications and ALSA PCM core. Once decided, runtime of +the PCM substream keeps the parameters. + +The parameters are described in :c:type:`struct snd_pcm_hw_params`. This +structure includes several types of parameters. Applications set preferable +value to these parameters, then execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE +or SNDRV_PCM_IOCTL_HW_PARAMS. The former is used just for refining available +set of parameters. The latter is used for an actual decision of the parameters. + +The :c:type:`struct snd_pcm_hw_params` structure has below members: + +``flags`` + Configurable. ALSA PCM core and some drivers handle this flag to select + convenient parameters or change their behaviour. +``masks`` + Configurable. This type of parameter is described in + :c:type:`struct snd_mask` and represent mask values. As of PCM protocol + v2.0.13, three types are defined. + + - SNDRV_PCM_HW_PARAM_ACCESS + - SNDRV_PCM_HW_PARAM_FORMAT + - SNDRV_PCM_HW_PARAM_SUBFORMAT +``intervals`` + Configurable. This type of parameter is described in + :c:type:`struct snd_interval` and represent values with a range. As of + PCM protocol v2.0.13, twelve types are defined. + + - SNDRV_PCM_HW_PARAM_SAMPLE_BITS + - SNDRV_PCM_HW_PARAM_FRAME_BITS + - SNDRV_PCM_HW_PARAM_CHANNELS + - SNDRV_PCM_HW_PARAM_RATE + - SNDRV_PCM_HW_PARAM_PERIOD_TIME + - SNDRV_PCM_HW_PARAM_PERIOD_SIZE + - SNDRV_PCM_HW_PARAM_PERIOD_BYTES + - SNDRV_PCM_HW_PARAM_PERIODS + - SNDRV_PCM_HW_PARAM_BUFFER_TIME + - SNDRV_PCM_HW_PARAM_BUFFER_SIZE + - SNDRV_PCM_HW_PARAM_BUFFER_BYTES + - SNDRV_PCM_HW_PARAM_TICK_TIME +``rmask`` + Configurable. This is evaluated at ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE only. Applications can select which + mask/interval parameter can be changed by ALSA PCM core. For + SNDRV_PCM_IOCTL_HW_PARAMS, this mask is ignored and all of parameters + are going to be changed. +``cmask`` + Read-only. After returning from ioctl(2), buffer in user space for + :c:type:`struct snd_pcm_hw_params` includes result of each operation. + This mask represents which mask/interval parameter is actually changed. +``info`` + Read-only. This represents hardware/driver capabilities as bit flags + with SNDRV_PCM_INFO_XXX. Typically, applications execute ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE to retrieve this flag, then decide candidates + of parameters and execute ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS to + configure PCM substream. +``msbits`` + Read-only. This value represents available bit width in MSB side of + a PCM sample. When a parameter of SNDRV_PCM_HW_PARAM_SAMPLE_BITS was + decided as a fixed number, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_num`` + Read-only. This value represents numerator of sampling rate in fraction + notation. Basically, when a parameter of SNDRV_PCM_HW_PARAM_RATE was + decided as a single value, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_den`` + Read-only. This value represents denominator of sampling rate in + fraction notation. Basically, when a parameter of + SNDRV_PCM_HW_PARAM_RATE was decided as a single value, this value is + also calculated according to it. Else, zero. But this behaviour depends + on implementations in driver side. +``fifo_size`` + Read-only. This value represents the size of FIFO in serial sound + interface of hardware. Basically, each driver can assigns a proper + value to this parameter but some drivers intentionally set zero with + a care of hardware design or data transmission protocol. + +ALSA PCM core handles buffer of :c:type:`struct snd_pcm_hw_params` when +applications execute ioctl(2) with SNDRV_PCM_HW_REFINE or SNDRV_PCM_HW_PARAMS. +Parameters in the buffer are changed according to +:c:type:`struct snd_pcm_hardware` and rules of constraints in the runtime. The +structure describes capabilities of handled hardware. The rules describes +dependencies on which a parameter is decided according to several parameters. +A rule has a callback function, and drivers can register arbitrary functions +to compute the target parameter. ALSA PCM core registers some rules to the +runtime as a default. + +Each driver can join in the interaction as long as it prepared for two stuffs +in a callback of :c:type:`struct snd_pcm_ops.open`. + +1. In the callback, drivers are expected to change a member of + :c:type:`struct snd_pcm_hardware` type in the runtime, according to + capacities of corresponding hardware. +2. In the same callback, drivers are also expected to register additional rules + of constraints into the runtime when several parameters have dependencies + due to hardware design. + +The driver can refers to result of the interaction in a callback of +:c:type:`struct snd_pcm_ops.hw_params`, however it should not change the +content. + +Tracepoints in this category are designed to trace changes of the +mask/interval parameters. When ALSA PCM core changes them, ``hw_mask_param`` or +``hw_interval_param`` event is probed according to type of the changed parameter. + +ALSA PCM core also has a pretty print format for each of the tracepoints. Below +is an example for ``hw_mask_param``. + +:: + + hw_mask_param: pcmC0D0p 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044 + + +Below is an example for ``hw_interval_param``. + +:: + + hw_interval_param: pcmC0D0p 000/023 BUFFER_SIZE 0 0 [0 4294967295] 0 1 [0 4294967295] + +The first three fields are common. They represent name of ALSA PCM character +device, rules of constraint and name of the changed parameter, in order. The +field for rules of constraint consists of two sub-fields; index of applied rule +and total number of rules added to the runtime. As an exception, the index 000 +means that the parameter is changed by ALSA PCM core, regardless of the rules. + +The rest of field represent state of the parameter before/after changing. These +fields are different according to type of the parameter. For parameters of mask +type, the fields represent hexadecimal dump of content of the parameter. For +parameters of interval type, the fields represent values of each member of +``empty``, ``integer``, ``openmin``, ``min``, ``max``, ``openmax`` in +:c:type:`struct snd_interval` in this order. + +Tracepoints in drivers +====================== + +Some drivers have tracepoints for developers' convenience. For them, please +refer to each documentation or implementation. -- cgit v1.2.3