diff options
Diffstat (limited to 'Documentation/devicetree/bindings/sound')
39 files changed, 1244 insertions, 397 deletions
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt index e79aeef73f28..9225472c80b4 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -17,12 +17,18 @@ Required properties: Documentation/devicetree/bindings/regulator/regulator.txt Optional properties: - - reset-gpio: the reset pin for the chip, for more details consult + - reset-gpios: the reset pin for the chip, for more details consult Documentation/devicetree/bindings/gpio/gpio.txt - DVDD-supply: supply voltage for the digital core, please consult Documentation/devicetree/bindings/regulator/regulator.txt +- adi,micbias: configures the voltage setting for the MICBIAS pin. + Select 0/1/2/3/4/5/6/7/8 to specify MICBIAS voltage + 5V/5.5V/6V/6.5V/7V/7.5V/8V/8.5V/9V + If not specified the default value will be "7" meaning 8.5 Volts. + This property is only valid for the ADAU1977 + For required properties on SPI, please consult Documentation/devicetree/bindings/spi/spi-bus.txt @@ -40,7 +46,8 @@ Examples: AVDD-supply = <®ulator>; DVDD-supply = <®ulator_digital>; - reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + adi,micbias = <3>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; }; adau1977_i2c: adau1977@11 { @@ -50,5 +57,5 @@ Examples: AVDD-supply = <®ulator>; DVDD-supply = <®ulator_digital>; - reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; }; diff --git a/Documentation/devicetree/bindings/sound/ak4104.txt b/Documentation/devicetree/bindings/sound/ak4104.txt index deca5e18f304..ae5f7f057dc3 100644 --- a/Documentation/devicetree/bindings/sound/ak4104.txt +++ b/Documentation/devicetree/bindings/sound/ak4104.txt @@ -12,8 +12,8 @@ Required properties: Optional properties: - - reset-gpio : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the device starts. + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the device starts. Example: diff --git a/Documentation/devicetree/bindings/sound/ak4118.txt b/Documentation/devicetree/bindings/sound/ak4118.txt new file mode 100644 index 000000000000..6e11a2f7404c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4118.txt @@ -0,0 +1,22 @@ +AK4118 S/PDIF transceiver + +This device supports I2C mode. + +Required properties: + +- compatible : "asahi-kasei,ak4118" +- reg : The I2C address of the device for I2C +- reset-gpios: A GPIO specifier for the reset pin +- irq-gpios: A GPIO specifier for the IRQ pin + +Example: + +&i2c { + ak4118: ak4118@13 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4118"; + reg = <0x13>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW> + irq-gpios = <&gpio 1 GPIO_ACTIVE_HIGH>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index 7839be78448d..e5820235e0d5 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -4,7 +4,7 @@ This device supports I2C mode. Required properties: -- compatible : "asahi-kasei,ak4458" +- compatible : "asahi-kasei,ak4458" or "asahi-kasei,ak4497" - reg : The I2C address of the device for I2C Optional properties: diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt new file mode 100644 index 000000000000..2e6cb7d9b202 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt @@ -0,0 +1,22 @@ +* Amlogic Audio SPDIF Input + +Required properties: +- compatible: 'amlogic,axg-spdifin' +- interrupts: interrupt specifier for the spdif input. +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "refclk" : spdif input reference clock +- #sound-dai-cells: must be 0. + +Example on the A113 SoC: + +spdifin: audio-controller@400 { + compatible = "amlogic,axg-spdifin"; + reg = <0x0 0x400 0x0 0x30>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>; + clocks = <&clkc_audio AUD_CLKID_SPDIFIN>, + <&clkc_audio AUD_CLKID_SPDIFIN_CLK>; + clock-names = "pclk", "refclk"; +}; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt index 7e63e53a901c..269682619a70 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -32,7 +32,9 @@ Required properties: Optional properties: - pa-gpios: GPIO used to control external amplifier. +----------------------- Example: Single DAI case +----------------------- sound_card { compatible = "audio-graph-card"; @@ -61,7 +63,9 @@ Example: Single DAI case }; }; +----------------------- Example: Multi DAI case +----------------------- sound-card { compatible = "audio-graph-card"; @@ -130,3 +134,204 @@ Example: Multi DAI case }; }; + +----------------------- +Example: Sampling Rate Conversion +----------------------- + + sound_card { + compatible = "audio-graph-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "DAI0 Capture", "codec Capture"; + 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"; + ... + }; + }; + }; + +----------------------- +Example: 2 CPU 1 Codec (Mixing) +----------------------- + + sound_card { + compatible = "audio-graph-card"; + + label = "sound-card"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback", + "DAI0 Capture", "codec Capture"; + + dais = <&cpu_port>; + }; + + audio-codec { + ... + + audio-graph-card,prefix = "codec"; + audio-graph-card,convert-rate = <48000>; + port { + reg = <0>; + codec_endpoint0: endpoint@0 { + remote-endpoint = <&cpu_endpoint0>; + }; + codec_endpoint1: endpoint@1 { + remote-endpoint = <&cpu_endpoint1>; + }; + }; + }; + + dai-controller { + ... + cpu_port: port { + cpu_endpoint0: endpoint@0 { + remote-endpoint = <&codec_endpoint0>; + + dai-format = "left_j"; + ... + }; + cpu_endpoint1: endpoint@1 { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + ... + }; + }; + }; + +----------------------- +Example: Multi DAI with DPCM +----------------------- + + CPU0 ------ ak4613 + CPU1 ------ HDMI + CPU2 ------ PCM3168A-p /* DPCM 1ch/2ch */ + CPU3 --/ /* DPCM 3ch/4ch */ + CPU4 --/ /* DPCM 5ch/6ch */ + CPU5 --/ /* DPCM 7ch/8ch */ + CPU6 ------ PCM3168A-c + + sound_card: sound { + compatible = "audio-graph-card"; + + label = "sound-card"; + + routing = "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback", + "pcm3168a Playback", "DAI5 Playback"; + + dais = <&snd_port0 /* ak4613 */ + &snd_port1 /* HDMI0 */ + &snd_port2 /* pcm3168a playback */ + &snd_port3 /* pcm3168a capture */ + >; + }; + + ak4613: codec@10 { + ... + port { + ak4613_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + }; + + pcm3168a: audio-codec@44 { + ... + audio-graph-card,prefix = "pcm3168a"; + audio-graph-card,convert-channels = <8>; /* TDM Split */ + ports { + port@0 { + reg = <0>; + pcm3168a_endpoint_p1: endpoint@1 { + remote-endpoint = <&rsnd_endpoint2>; + ... + }; + pcm3168a_endpoint_p2: endpoint@2 { + remote-endpoint = <&rsnd_endpoint3>; + ... + }; + pcm3168a_endpoint_p3: endpoint@3 { + remote-endpoint = <&rsnd_endpoint4>; + ... + }; + pcm3168a_endpoint_p4: endpoint@4 { + remote-endpoint = <&rsnd_endpoint5>; + ... + }; + }; + port@1 { + reg = <1>; + pcm3168a_endpoint_c: endpoint { + remote-endpoint = <&rsnd_endpoint6>; + ... + }; + }; + }; + }; + + &sound { + ports { + snd_port0: port@0 { + rsnd_endpoint0: endpoint { + remote-endpoint = <&ak4613_endpoint>; + ... + }; + }; + snd_port1: port@1 { + rsnd_endpoint1: endpoint { + remote-endpoint = <&dw_hdmi0_snd_in>; + ... + }; + }; + snd_port2: port@2 { + #address-cells = <1>; + #size-cells = <0>; + rsnd_endpoint2: endpoint@2 { + remote-endpoint = <&pcm3168a_endpoint_p1>; + ... + }; + rsnd_endpoint3: endpoint@3 { + remote-endpoint = <&pcm3168a_endpoint_p2>; + ... + }; + rsnd_endpoint4: endpoint@4 { + remote-endpoint = <&pcm3168a_endpoint_p3>; + ... + }; + rsnd_endpoint5: endpoint@5 { + remote-endpoint = <&pcm3168a_endpoint_p4>; + ... + }; + }; + snd_port3: port@6 { + rsnd_endpoint6: endpoint { + remote-endpoint = <&pcm3168a_endpoint_c>; + ... + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt deleted file mode 100644 index 441dd6f29df1..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ /dev/null @@ -1,123 +0,0 @@ -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 1. Sampling Rate Conversion - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - 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"; - ... - }; - }; - }; - -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", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port0 - &cpu_port1>; - }; - - audio-codec { - ... - - port { - codec_endpoint0: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 000000000000..912bd162b477 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0x00>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt index 6b222f9b8ef5..c33770ec4c3c 100644 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -10,8 +10,8 @@ Required properties: Optional properties: - - reset-gpio : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the codec starts. + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. Example: diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt new file mode 100644 index 000000000000..12b4aa8ef0db --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -0,0 +1,22 @@ +Cirrus Logic CS4341 audio DAC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "cirrus,cs4341a" + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +For required properties on I2C-bus, please consult +Documentation/devicetree/bindings/i2c/i2c.txt +For required properties on SPI-bus, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Example: + codec: cs4341@0 { + #sound-dai-cells = <0>; + compatible = "cirrus,cs4341a"; + reg = <0>; + spi-max-frequency = <6000000>; + }; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 46bc9829c71a..a58f79f5345c 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -30,6 +30,11 @@ Optional properties: - ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 - tx-num-evt : FIFO levels. - rx-num-evt : FIFO levels. +- dismod : Specify the drive on TX pin during inactive slots + 0 : 3-state + 2 : logic low + 3 : logic high + Defaults to 'logic low' when the property is not present - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture - interrupts : Interrupt numbers for McASP @@ -40,6 +45,23 @@ Optional properties: - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +Optional GPIO support: +If any McASP pin need to be used as GPIO then the McASP node must have: +... + gpio-controller + #gpio-cells = <2>; +... + +When requesting a GPIO, the first parameter is the PIN index in McASP_P* +registers. +For example to request the AXR2 pin of mcasp8: +function-gpios = <&mcasp8 2 0>; + +Or to request the ACLKR pin of mcasp8: +function-gpios = <&mcasp8 29 0>; + +For generic gpio information, please refer to bindings/gpio/gpio.txt + Example: mcasp0: mcasp0@1d00000 { diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index e957b4136716..32e871037269 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -9,6 +9,7 @@ Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop - num-channels: Number of microphones on this DAI - wakeup-delay-ms: Delay (in ms) after enabling the DMIC + - modeswitch-delay-ms: Delay (in ms) to complete DMIC mode switch Example node: @@ -17,4 +18,5 @@ Example node: dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; num-channels = <1>; wakeup-delay-ms <50>; + modeswitch-delay-ms <35>; }; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.txt b/Documentation/devicetree/bindings/sound/fsl,micfil.txt new file mode 100644 index 000000000000..53e227b15277 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.txt @@ -0,0 +1,32 @@ +NXP MICFIL Digital Audio Interface (MICFIL). + +The MICFIL digital interface provides a 16-bit audio signal from a PDM +microphone bitstream in a configurable output sampling rate. + +Required properties: + + - compatible : Compatible list, contains "fsl,imx8mm-micfil" + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the micfil interrupts. + + - clocks : Must contain an entry for each entry in clock-names. + + - clock-names : Must include the "ipg_clk" for register access and + "ipg_clk_app" for internal micfil clock. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + +Example: +micfil: micfil@30080000 { + compatible = "fsl,imx8mm-micfil"; + reg = <0x0 0x30080000 0x0 0x10000>; + interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>; + clock-names = "ipg_clk", "ipg_clk_app"; + dmas = <&sdma2 24 26 0x80000000>; +}; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index dd9e59738e08..2e726b983845 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -35,13 +35,13 @@ Required properties: - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transimitter and the + with Rx) which means both the transmitter and the receiver will send and receive data by following receiver's bit clocks and frame sync clocks. - fsl,sai-asynchronous: This is a boolean property. If present, indicating that SAI will work in the asynchronous mode, which - means both transimitter and receiver will send and + means both transmitter and receiver will send and receive data by following their own bit clocks and frame sync clocks separately. @@ -58,8 +58,8 @@ Optional properties (for mx6ul): Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both - transimitter and receiver will send and receive data by following clocks - of transimitter. + transmitter and receiver will send and receive data by following clocks + of transmitter. - fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. Example: diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt new file mode 100644 index 000000000000..1084f7f22eea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt @@ -0,0 +1,26 @@ +* Audio codec controlled by ChromeOS EC + +Google's ChromeOS EC codec is a digital mic codec provided by the +Embedded Controller (EC) and is controlled via a host-command interface. + +An EC codec node should only be found as a sub-node of the EC node (see +Documentation/devicetree/bindings/mfd/cros-ec.txt). + +Required properties: +- compatible: Must contain "google,cros-ec-codec" +- #sound-dai-cells: Should be 1. The cell specifies number of DAIs. +- max-dmic-gain: A number for maximum gain in dB on digital microphone. + +Example: + +cros-ec@0 { + compatible = "google,cros-ec-spi"; + + ... + + cros_ec_codec: ec-codec { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + max-dmic-gain = <43>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt new file mode 100644 index 000000000000..05adc0d47b13 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4725B codec controller + +Required properties: +- compatible : "ingenic,jz4725b-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@100200a4 { + compatible = "ingenic,jz4725b-codec"; + reg = <0x100200a4 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4725B_CLK_AIC>; + clock-names = "aic"; +}; diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt new file mode 100644 index 000000000000..1ffcade87e7b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4740 codec controller + +Required properties: +- compatible : "ingenic,jz4740-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; +}; diff --git a/Documentation/devicetree/bindings/sound/mt6358.txt b/Documentation/devicetree/bindings/sound/mt6358.txt new file mode 100644 index 000000000000..5465730013a1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6358.txt @@ -0,0 +1,18 @@ +Mediatek MT6358 Audio Codec + +The communication between MT6358 and SoC is through Mediatek PMIC wrapper. +For more detail, please visit Mediatek PMIC wrapper documentation. + +Must be a child node of PMIC wrapper. + +Required properties: + +- compatible : "mediatek,mt6358-sound". +- Avdd-supply : power source of AVDD + +Example: + +mt6358_snd { + compatible = "mediatek,mt6358-sound"; + Avdd-supply = <&mt6358_vaud28_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt new file mode 100644 index 000000000000..396ba38619f6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt @@ -0,0 +1,36 @@ +Mediatek AFE PCM controller for mt8183 + +Required properties: +- compatible = "mediatek,mt68183-audio"; +- reg: register location and size +- interrupts: should contain AFE interrupt +- power-domains: should define the power domain +- clocks: Must contain an entry for each entry in clock-names +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_clk26m_clk"; + +Example: + + afe: mt8183-afe-pcm@11220000 { + compatible = "mediatek,mt8183-audio"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>; + power-domains = <&scpsys MT8183_POWER_DOMAIN_AUDIO>; + clocks = <&infrasys CLK_INFRA_AUDIO>, + <&infrasys CLK_INFRA_AUDIO_26M_BCLK>, + <&topckgen CLK_TOP_MUX_AUDIO>, + <&topckgen CLK_TOP_MUX_AUD_INTBUS>, + <&topckgen CLK_TOP_SYSPLL_D2_D4>, + <&clk26m>; + clock-names = "infra_sys_audio_clk", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll_d2_d4", + "top_clk26m_clk"; + }; diff --git a/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt new file mode 100644 index 000000000000..679e44839b48 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt @@ -0,0 +1,24 @@ +Mediatek ALSA BT SCO CVSD/MSBC Driver + +Required properties: +- compatible = "mediatek,mtk-btcvsd-snd"; +- reg: register location and size of PKV and SRAM_BANK2 +- interrupts: should contain BTSCO interrupt +- mediatek,infracfg: the phandles of INFRASYS +- mediatek,offset: Array contains of register offset and mask + infra_misc_offset, + infra_conn_bt_cvsd_mask, + cvsd_mcu_read_offset, + cvsd_mcu_write_offset, + cvsd_packet_indicator_offset + +Example: + + mtk-btcvsd-snd@18000000 { + compatible = "mediatek,mtk-btcvsd-snd"; + reg=<0 0x18000000 0 0x1000>, + <0 0x18080000 0 0x8000>; + interrupts = <GIC_SPI 286 IRQ_TYPE_LEVEL_LOW>; + mediatek,infracfg = <&infrasys>; + mediatek,offset = <0xf00 0x800 0xfd0 0xfd4 0xfd8>; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt index 44d27456e8a4..21cd310963b1 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -13,6 +13,10 @@ Required properties: See ../reset/reset.txt for details. - reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x +Optional properties: +- nvidia,model : The user-visible name of this sound complex. Since the property + is optional, legacy boards can use default name provided in hda driver. + Example: hda@70030000 { @@ -27,4 +31,5 @@ hda@70030000 { <&tegra_car 128>, /* hda2hdmi */ <&tegra_car 111>; /* hda2codec_2x */ reset-names = "hda", "hda2hdmi", "hda2codec_2x"; + nvidia,model = "jetson-tk1-hda"; }; diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt index 5f4e68ca228c..ff98a0cb5b3f 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt @@ -7,6 +7,8 @@ Required properties: <L3 interconnect address, size>; - interrupts: Interrupt number for McPDM - ti,hwmods: Name of the hwmod associated to the McPDM +- clocks: phandle for the pdmclk provider, likely <&twl6040> +- clock-names: Must be "pdmclk" Example: @@ -18,3 +20,11 @@ mcpdm: mcpdm@40132000 { interrupt-parent = <&gic>; ti,hwmods = "mcpdm"; }; + +In board DTS file the pdmclk needs to be added: + +&mcpdm { + clocks = <&twl6040>; + clock-names = "pdmclk"; + status = "okay"; +}; diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt index 90fcb8523099..97de66932d44 100644 --- a/Documentation/devicetree/bindings/sound/pcm3060.txt +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -9,9 +9,15 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + Examples: pcm3060: pcm3060@46 { compatible = "ti,pcm3060"; reg = <0x46>; + ti,out-single-ended = "true"; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt index fdcea3d12ee5..e7d17dda55db 100644 --- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt +++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt @@ -30,6 +30,7 @@ Required properties - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node. - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node. - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node. + Optional Properties: - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons detection on headset when the mbhc is powered up @@ -92,9 +93,9 @@ spmi_bus { "cdc_ear_cnp_int", "cdc_hphr_cnp_int", "cdc_hphl_cnp_int"; - VDD-CDC-IO-supply = <&pm8916_l5>; - VDD-CDC-TX-RX-CX-supply = <&pm8916_l5>; - VDD-MICBIAS-supply = <&pm8916_l13>; + vdd-cdc-io-supply = <&pm8916_l5>; + vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>; + vdd-micbias-supply = <&pm8916_l13>; #sound-dai-cells = <1>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index f9c7bd8c1bc0..9f5378c51686 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -27,6 +27,28 @@ used by the apr service device. Value type: <u32> Definition: Must be 1 +== ASM DAI is subnode of "dais" and represent a dai, it includes board specific +configuration of each dai. Must contain the following properties. + +- reg + Usage: required + Value type: <u32> + Definition: Must be dai id + +- direction: + Usage: Required for Compress offload dais + Value type: <u32> + Definition: Specifies the direction of the dai stream + 0 for both tx and rx + 1 for only tx (Capture/Encode) + 2 for only rx (Playback/Decode) + +- is-compress-dai: + Usage: Required for Compress offload dais + Value type: <boolean> + Definition: present for Compress offload dais + + = EXAMPLE q6asm@7 { @@ -35,5 +57,10 @@ q6asm@7 { q6asmdai: dais { compatible = "qcom,q6asm-dais"; #sound-dai-cells = <1>; + mm@0 { + reg = <0>; + direction = <2>; + is-compress-dai; + }; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt index 1d8d49e30af7..5d6ea66a863f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt +++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt @@ -34,12 +34,12 @@ Required properties with SLIMbus Interface: Definition: Interrupt names of WCD INTR1 and INTR2 Should be: "intr1", "intr2" -- reset-gpio: +- reset-gpios: Usage: required Value type: <String Array> Definition: Reset gpio line -- qcom,ifd: +- slim-ifc-dev: Usage: required Value type: <phandle> Definition: SLIM interface device @@ -104,13 +104,13 @@ Required properties with SLIMbus Interface: Value type: <u32> Definition: Must be 1 -codec@1{ +audio-codec@1{ compatible = "slim217,1a0"; reg = <1 0>; interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>; interrupt-names = "intr2" - reset-gpio = <&msmgpio 64 0>; - qcom,ifd = <&wc9335_ifd>; + reset-gpios = <&msmgpio 64 0>; + slim-ifc-dev = <&wc9335_ifd>; clock-names = "mclk", "native"; clocks = <&rpmcc RPM_SMD_DIV_CLK1>, <&rpmcc RPM_SMD_BB_CLK1>; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index d92b705e7917..648d43e1b1e9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -39,15 +39,7 @@ This is example of Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] - &rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; - }; +see "Example: simple sound card" You can use below. ${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. @@ -83,29 +75,8 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "simple-scu-audio-card" sound card for it. -example) - - sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; - }; +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +see "Example: simple sound card for Asynchronous mode" ------------------ ** Synchronous mode @@ -141,26 +112,8 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "simple-scu-audio-card" sound card for it. -example) - - sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; - }; +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +see "Example: simple sound card for channel convert" Ex) Exchange output channel Input -> Output @@ -190,42 +143,13 @@ 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 "simple-scu-audio-card" sound card for it. +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ - sound { - #address-cells = <1>; - #size-cells = <0>; - - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - reg = <0>; - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - reg = <1>; - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; - }; - - &rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; - }; +see "Example: simple sound card for MIXer" ============================================= * DVC (Digital Volume and Mute Function) @@ -257,15 +181,31 @@ Volume Ramp * SSIU (Serial Sound Interface Unit) ============================================= -There is no DT settings for SSIU, because SSIU will be automatically -selected via SSI. SSIU can avoid some under/over run error, because it has some buffer. But you can't use it if SSI was PIO mode. -In DMA mode, you can select not to use SSIU by using "no-busif" on DT. +In DMA mode, you can select not to use SSIU by using "no-busif" via SSI. - &ssi0 { - no-busif; - }; +SSIU handles BUSIF which will be used for TDM Split mode. +This driver is assuming that audio-graph card will be used. + +TDM Split mode merges 4 sounds. You can see 4 sound interface on system, +and these sounds will be merged SSIU/SSI. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 xxxx.wav & + aplay -D plughw:0,2 xxxx.wav & + aplay -D plughw:0,3 xxxx.wav + + 2ch 8ch + [MEM] -> [SSIU 30] -+-> [SSIU 3] --> [Codec] + 2ch | + [MEM] -> [SSIU 31] -+ + 2ch | + [MEM] -> [SSIU 32] -+ + 2ch | + [MEM] -> [SSIU 33] -+ + +see "Example: simple sound card for TDM Split" ============================================= * SSI (Serial Sound Interface) @@ -304,14 +244,7 @@ This is example if SSI1 want to share WS pin with SSI0 You can use Multi-SSI. This is example of SSI0/SSI1/SSI2 (= for 6ch) - &rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; - }; +see "Example: simple sound card for Multi channel" ** TDM-SSI @@ -319,19 +252,7 @@ You can use TDM with SSI. This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. - rsnd_tdm: sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,cpu { - /* system can use TDM 6ch */ - dai-tdm-slot-num = <6>; - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; - }; - +see "Example: simple sound card for TDM" ============================================= Required properties: @@ -346,6 +267,7 @@ Required properties: - "renesas,rcar_sound-r8a7744" (RZ/G1N) - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) + - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7779" (R-Car H1) - "renesas,rcar_sound-r8a7790" (R-Car H2) @@ -356,6 +278,7 @@ Required properties: - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - "renesas,rcar_sound-r8a77990" (R-Car E3) + - "renesas,rcar_sound-r8a77995" (R-Car D3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 @@ -363,6 +286,9 @@ Required properties: - rcar_sound,ssi : Should contain SSI feature. The number of SSI subnode should be same as HW. see below for detail. +- rcar_sound,ssiu : Should contain SSIU feature. + The number of SSIU subnode should be same as HW. + see below for detail. - rcar_sound,src : Should contain SRC feature. The number of SRC subnode should be same as HW. see below for detail. @@ -402,8 +328,13 @@ SSI subnode properties: - no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case - dma : Should contain Audio DMAC entry - dma-names : SSI case "rx" (=playback), "tx" (=capture) + Deprecated: see SSIU subnode properties SSIU case "rxu" (=playback), "txu" (=capture) +SSIU subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "rx" (=playback), "tx" (=capture) + SRC subnode properties: - dma : Should contain Audio DMAC entry - dma-names : "rx" (=playback), "tx" (=capture) @@ -532,56 +463,55 @@ rcar_sound: sound@ec500000 { }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + + ... + + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; }; ssi1: ssi-1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi2: ssi-2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi3: ssi-3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi4: ssi-4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi5: ssi-5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi6: ssi-6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi7: ssi-7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; }; + + ... + ssi8: ssi-8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x11>, <&audma1 0x12>; + dma-names = "rx", "tx"; }; ssi9: ssi-9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x13>, <&audma1 0x14>; + dma-names = "rx", "tx"; }; }; @@ -647,25 +577,174 @@ Example: simple sound card }; ============================================= +Example: simple sound card for Asynchronous mode +============================================= + +sound { + compatible = "simple-scu-audio-card"; + ... + /* + * SRC Asynchronous mode setting + * Playback: + * All input data will be converted to 48kHz + * Capture: + * Inputed 48kHz data will be converted to + * system specified Hz + */ + simple-audio-card,convert-rate = <48000>; + ... + simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; +}; + +============================================= +Example: simple sound card for channel convert +============================================= + +sound { + compatible = "simple-scu-audio-card"; + ... + /* + * CTU setting + * All input data will be converted to 2ch + * as output data + */ + simple-audio-card,convert-channels = <2>; + ... + simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; +}; + +============================================= +Example: simple sound card for MIXer +============================================= + +sound { + compatible = "simple-scu-audio-card"; + ... + simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + simple-audio-card,codec { + ... + }; +}; + +&rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; + }; + dai1 { + playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; + }; + }; +}; + +============================================= Example: simple sound card for TDM ============================================= - rsnd_tdm: sound { - compatible = "simple-audio-card"; +rsnd_tdm: sound { + compatible = "simple-audio-card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + dai-tdm-slot-num = <6>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&xxx>; + }; +}; + +============================================= +Example: simple sound card for TDM Split +============================================= + +sound_card: sound { + compatible = "audio-graph-scu-card"; + prefix = "xxxx"; + routing = "xxxx Playback", "DAI0 Playback", + "xxxx Playback", "DAI1 Playback", + "xxxx Playback", "DAI2 Playback", + "xxxx Playback", "DAI3 Playback"; + convert-channels = <8>; /* TDM Split */ + + dais = <&rsnd_port0 /* playback ch1/ch2 */ + &rsnd_port1 /* playback ch3/ch4 */ + &rsnd_port2 /* playback ch5/ch6 */ + &rsnd_port3 /* playback ch7/ch8 */ + >; +}; + +audio-codec { + ... + port { + codec_0: endpoint@1 { + remote-endpoint = <&rsnd_ep0>; + }; + codec_1: endpoint@2 { + remote-endpoint = <&rsnd_ep1>; + }; + codec_2: endpoint@3 { + remote-endpoint = <&rsnd_ep2>; + }; + codec_3: endpoint@4 { + remote-endpoint = <&rsnd_ep3>; }; + }; +}; - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; +&rcar_sound { + ... + ports { + rsnd_port0: port@0 { + rsnd_ep0: endpoint { + remote-endpoint = <&codec_0>; + ... + playback = <&ssiu30 &ssi3>; + }; + }; + rsnd_port1: port@1 { + rsnd_ep1: endpoint { + remote-endpoint = <&codec_1>; + ... + playback = <&ssiu31 &ssi3>; + }; + }; + rsnd_port2: port@2 { + rsnd_ep2: endpoint { + remote-endpoint = <&codec_2>; + ... + playback = <&ssiu32 &ssi3>; + }; + }; + rsnd_port3: port@3 { + rsnd_ep3: endpoint { + remote-endpoint = <&codec_3>; + ... + playback = <&ssiu33 &ssi3>; + }; }; }; +}; ============================================= Example: simple sound card for Multi channel diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt new file mode 100644 index 000000000000..2469588c7ccb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -0,0 +1,23 @@ +* Rockchip Rk3328 internal codec + +Required properties: + +- compatible: "rockchip,rk3328-codec" +- reg: physical base address of the controller and length of memory mapped + region. +- rockchip,grf: the phandle of the syscon node for GRF register. +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should be "pclk". +- spk-depop-time-ms: speak depop time msec. + +Example for rk3328 internal codec: + +codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0x0 0xff410000 0x0 0x1000>; + rockchip,grf = <&grf>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + spk-depop-time-ms = 100; + status = "disabled"; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt index 92b986ca337b..56bc85232c49 100644 --- a/Documentation/devicetree/bindings/sound/rt5631.txt +++ b/Documentation/devicetree/bindings/sound/rt5631.txt @@ -35,14 +35,14 @@ Pins on the device (for linking into audio routes): Example: -alc5631: alc5631@1a { +alc5631: audio-codec@1a { compatible = "realtek,alc5631"; reg = <0x1a>; }; or -rt5631: rt5631@1a { +rt5631: audio-codec@1a { compatible = "realtek,rt5631"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d..2a55e9133408 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -10,6 +10,10 @@ Required properties: - interrupts : The CODEC's interrupt output. +- avdd-supply: Power supply for AVDD, providing 1.8V. + +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Optional properties: - "realtek,dc_offset_l_manual" @@ -51,4 +55,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 9c58f724396a..9d9ff5184939 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -37,6 +37,15 @@ VDDIO 1.8V 2.5V 3.3V 2 = 3.33 mA 5.74 mA 8.03 mA 3 = 4.99 mA 8.61 mA 12.05 mA +- sclk-strength: the SCLK pad strength. Possible values are: +0, 1, 2 and 3 as per the table below: + +VDDIO 1.8V 2.5V 3.3V +0 = Disable +1 = 1.66 mA 2.87 mA 4.02 mA +2 = 3.33 mA 5.74 mA 8.03 mA +3 = 4.99 mA 8.61 mA 12.05 mA + Example: sgtl5000: codec@a { diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt index 8647edae7af0..7182ac4f1e65 100644 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -4,9 +4,14 @@ Required properties: - compatible : "dioo,dio2125" or "simple-audio-amplifier" - enable-gpios : the gpio connected to the enable pin of the simple amplifier +Optional properties: +- VCC-supply : power supply for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + Example: amp: analog-amplifier { compatible = "simple-audio-amplifier"; + VCC-supply = <®ulator>; enable-gpios = <&gpio GPIOH_3 0>; }; diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index a4c72d09cd45..4629c8f8a6b6 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -95,7 +95,9 @@ Optional CPU/CODEC subnodes properties: initialization. It is useful for some aCPUs with fixed clocks. +------------------------------------------- Example 1 - single DAI link: +------------------------------------------- sound { compatible = "simple-audio-card"; @@ -138,7 +140,9 @@ sh_fsi2: sh_fsi2@ec230000 { interrupts = <0 146 0x4>; }; +------------------------------------------- Example 2 - many DAI links: +------------------------------------------- sound { compatible = "simple-audio-card"; @@ -176,8 +180,10 @@ sound { }; }; +------------------------------------------- Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec through TPA6130A2 amplifier to headphones: +------------------------------------------- &i2c0 { codec: tlv320dac3100@18 { @@ -210,3 +216,134 @@ sound { clocks = ... }; }; + +------------------------------------------- +Example 4. Sampling Rate Conversion +------------------------------------------- + +sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; +}; + +------------------------------------------- +Example 5. 2 CPU 1 Codec (Mixing) +------------------------------------------- +sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; +}; + +------------------------------------------- +Example 6 - many DAI links with DPCM: +------------------------------------------- + +CPU0 ------ ak4613 +CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +CPU2 --/ /* DPCM 3ch/4ch */ +CPU3 --/ /* DPCM 5ch/6ch */ +CPU4 --/ /* DPCM 7ch/8ch */ +CPU5 ------ PCM3168A-c + +sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt deleted file mode 100644 index 32f8dbce5241..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt +++ /dev/null @@ -1,94 +0,0 @@ -ASoC Simple SCU Sound Card - -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 : 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 : 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 routing -- simple-audio-card,widgets : Please refer to widgets.txt. -- 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. - use audio-prefix if some components is using same sink/sources naming. - it can be used if compatible was "renesas,rsrc-card"; - -Required CPU/CODEC subnodes properties: - -- sound-dai : see simple-audio-card.txt - -Optional CPU/CODEC subnodes properties: - -- clocks / system-clock-frequency : see simple-audio-card.txt - -Example 1. Sampling Rate Conversion - -sound { - compatible = "simple-scu-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -Example 2. 2 CPU 1 Codec (Mixing) - -sound { - compatible = "simple-scu-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt new file mode 100644 index 000000000000..4b23e84b2e57 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd-pcm.txt @@ -0,0 +1,23 @@ +* Spreadtrum DMA platfrom bindings + +Required properties: +- compatible: Should be "sprd,pcm-platform". +- dmas: Specify the list of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + +Example: + + audio_platform:platform@0 { + compatible = "sprd,pcm-platform"; + dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, + <&agcp_dma 3 3>, <&agcp_dma 4 4>, + <&agcp_dma 5 5>, <&agcp_dma 6 6>, + <&agcp_dma 7 7>, <&agcp_dma 8 8>, + <&agcp_dma 9 9>, <&agcp_dma 10 10>; + dma-names = "normal_p_l", "normal_p_r", + "normal_c_l", "normal_c_r", + "voice_c", "fast_p", + "loop_c", "loop_p", + "voip_c", "voip_p"; + }; diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt index 4f8ad0e04d20..056a098495cc 100644 --- a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt +++ b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt @@ -4,9 +4,11 @@ Required properties: - compatible: must be one of the following compatibles: - "allwinner,sun50i-a64-codec-analog" - reg: must contain the registers location and length +- cpvdd-supply: Regulator supply for the headphone amplifier Example: codec_analog: codec-analog@1f015c0 { compatible = "allwinner,sun50i-a64-codec-analog"; reg = <0x01f015c0 0x4>; + cpvdd-supply = <®_eldo1>; }; diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000000..cbc93c8f4963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/xlnx,i2s.txt b/Documentation/devicetree/bindings/sound/xlnx,i2s.txt new file mode 100644 index 000000000000..5e7c7d5bb60a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,i2s.txt @@ -0,0 +1,28 @@ +Device-Tree bindings for Xilinx I2S PL block + +The IP supports I2S based playback/capture audio + +Required property: + - compatible: "xlnx,i2s-transmitter-1.0" for playback and + "xlnx,i2s-receiver-1.0" for capture + +Required property common to both I2S playback and capture: + - reg: Base address and size of the IP core instance. + - xlnx,dwidth: sample data width. Can be any of 16, 24. + - xlnx,num-channels: Number of I2S streams. Can be any of 1, 2, 3, 4. + supported channels = 2 * xlnx,num-channels + +Example: + + i2s_receiver@a0080000 { + compatible = "xlnx,i2s-receiver-1.0"; + reg = <0x0 0xa0080000 0x0 0x10000>; + xlnx,dwidth = <0x18>; + xlnx,num-channels = <1>; + }; + i2s_transmitter@a0090000 { + compatible = "xlnx,i2s-transmitter-1.0"; + reg = <0x0 0xa0090000 0x0 0x10000>; + xlnx,dwidth = <0x18>; + xlnx,num-channels = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/xlnx,spdif.txt b/Documentation/devicetree/bindings/sound/xlnx,spdif.txt new file mode 100644 index 000000000000..15c2d64d247c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,spdif.txt @@ -0,0 +1,28 @@ +Device-Tree bindings for Xilinx SPDIF IP + +The IP supports playback and capture of SPDIF audio + +Required properties: + - compatible: "xlnx,spdif-2.0" + - clock-names: List of input clocks. + Required elements: "s_axi_aclk", "aud_clk_i" + - clocks: Input clock specifier. Refer to common clock bindings. + - reg: Base address and address length of the IP core instance. + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - xlnx,spdif-mode: 0 :- receiver mode + 1 :- transmitter mode + - xlnx,aud_clk_i: input audio clock value. + +Example: + spdif_0: spdif@80010000 { + clock-names = "aud_clk_i", "s_axi_aclk"; + clocks = <&misc_clk_0>, <&clk 71>; + compatible = "xlnx,spdif-2.0"; + interrupt-names = "spdif_interrupt"; + interrupt-parent = <&gic>; + interrupts = <0 91 4>; + reg = <0x0 0x80010000 0x0 0x10000>; + xlnx,spdif-mode = <1>; + xlnx,aud_clk_i = <49152913>; + }; |