diff options
Diffstat (limited to 'Documentation/devicetree/bindings/sound')
278 files changed, 15294 insertions, 4317 deletions
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1372.yaml b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml new file mode 100644 index 000000000000..59f7c60a14ba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1372.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + + +title: Analog Devices ADAU1372 CODEC + +maintainers: + - Alexandre Belloni <alexandre.belloni@bootlin.om> + +description: | + Analog Devices ADAU1372 four inputs and two outputs codec. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf + +properties: + compatible: + enum: + - adi,adau1372 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + powerdown-gpios: + description: GPIO used for hardware power-down. + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@3c { + compatible = "adi,adau1372"; + reg = <0x3c>; + #sound-dai-cells = <0>; + clock-names = "mclk"; + clocks = <&adau1372z_xtal>; + }; + }; + + adau1372z_xtal: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <12288000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt deleted file mode 100644 index 9225472c80b4..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt +++ /dev/null @@ -1,61 +0,0 @@ -Analog Devices ADAU1977/ADAU1978/ADAU1979 - -Datasheets: -http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf -http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf -http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf - -This driver supports both the I2C and SPI bus. - -Required properties: - - compatible: Should contain one of the following: - "adi,adau1977" - "adi,adau1978" - "adi,adau1979" - - - AVDD-supply: analog power supply for the device, please consult - Documentation/devicetree/bindings/regulator/regulator.txt - -Optional properties: - - 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 - -Required properties on I2C: - - - reg: The i2c address. Value depends on the state of ADDR0 - and ADDR1, as wired in hardware. - -Examples: - - adau1977_spi: adau1977@0 { - compatible = "adi,adau1977"; - spi-max-frequency = <600000>; - - AVDD-supply = <®ulator>; - DVDD-supply = <®ulator_digital>; - - adi,micbias = <3>; - reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; - }; - - adau1977_i2c: adau1977@11 { - compatible = "adi,adau1977"; - reg = <0x11>; - - AVDD-supply = <®ulator>; - DVDD-supply = <®ulator_digital>; - - reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml new file mode 100644 index 000000000000..b80454ad97da --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml @@ -0,0 +1,92 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1977.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1977/ADAU1978/ADAU1979 Quad ADC with Diagnostics + +maintainers: + - Lars-Peter Clausen <lars@metafoo.de> + - Bogdan Togorean <bogdan.togorean@analog.com> + +description: | + Analog Devices ADAU1977 and similar quad ADC with Diagnostics + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf + +properties: + compatible: + enum: + - adi,adau1977 + - adi,adau1978 + - adi,adau1979 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + reset-gpios: + maxItems: 1 + + spi-max-frequency: true + + AVDD-supply: + description: Analog power support for the device. + + DVDD-supply: + description: Supply voltage for digital core. + + adi,micbias: + description: | + Configures the voltage setting for the MICBIAS pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3, 4, 5, 6, 7, 8] + default: 7 + +required: + - reg + - compatible + - AVDD-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + spi { + #address-cells = <1>; + #size-cells = <0>; + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + reg = <0>; + spi-max-frequency = <600000>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + + adi,micbias = <3>; + }; + }; + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml new file mode 100644 index 000000000000..fb78967ee17b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7118.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + + +title: Analog Devices ADAU7118 8 Channel PDM to I2S/TDM Converter + +maintainers: + - Nuno Sá <nuno.sa@analog.com> + +description: | + Analog Devices ADAU7118 8 Channel PDM to I2S/TDM Converter over I2C or HW + standalone mode. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU7118.pdf + +properties: + compatible: + enum: + - adi,adau7118 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + iovdd-supply: + description: Digital Input/Output Power Supply. + + dvdd-supply: + description: Internal Core Digital Power Supply. + + adi,decimation-ratio: + description: | + This property set's the decimation ratio of PDM to PCM audio data. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [64, 32, 16] + default: 64 + + adi,pdm-clk-map: + description: | + The ADAU7118 has two PDM clocks for the four Inputs. Each input must be + assigned to one of these two clocks. This property set's the mapping + between the clocks and the inputs. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + items: + maximum: 1 + default: [0, 0, 1, 1] + +required: + - "#sound-dai-cells" + - compatible + - iovdd-supply + - dvdd-supply + +additionalProperties: false + +examples: + - | + i2c { + /* example with i2c support */ + #address-cells = <1>; + #size-cells = <0>; + adau7118_codec: audio-codec@14 { + compatible = "adi,adau7118"; + reg = <0x14>; + #sound-dai-cells = <0>; + iovdd-supply = <&supply>; + dvdd-supply = <&supply>; + adi,pdm-clk-map = <1 1 0 0>; + adi,decimation-ratio = <16>; + }; + }; + + /* example with hw standalone mode */ + adau7118_codec_hw: adau7118-codec-hw { + compatible = "adi,adau7118"; + #sound-dai-cells = <0>; + iovdd-supply = <&supply>; + dvdd-supply = <&supply>; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml new file mode 100644 index 000000000000..ec4c10c2598a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98396.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98396 Speaker Amplifier Device Tree Bindings + +maintainers: + - Ryan Lee <ryans.lee@analog.com> + +description: + The MAX98396 is a mono Class-DG speaker amplifier with I/V sense. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + The MAX98397 is a variant of MAX98396 with wide input supply range. + +properties: + compatible: + enum: + - adi,max98396 + - adi,max98397 + reg: + maxItems: 1 + description: I2C address of the device. + + adi,vmon-slot-no: + description: slot number of the voltage sense monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current sense monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,spkfb-slot-no: + description: slot number of speaker DSP monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V sense data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98396: amplifier@39 { + compatible = "adi,max98396"; + reg = <0x39>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4375.yaml b/Documentation/devicetree/bindings/sound/ak4375.yaml new file mode 100644 index 000000000000..f1d5074a024d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4375.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4375 DAC and headphones amplifier Device Tree Bindings + +maintainers: + - Vincent Knecht <vincent.knecht@mailoo.org> + +properties: + compatible: + const: asahi-kasei,ak4375 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + avdd-supply: + description: regulator phandle for the AVDD power supply. + + tvdd-supply: + description: regulator phandle for the TVDD power supply. + + pdn-gpios: + description: optional GPIO to set the PDN pin. + +required: + - compatible + - reg + - '#sound-dai-cells' + - avdd-supply + - tvdd-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + headphones: audio-codec@10 { + compatible = "asahi-kasei,ak4375"; + reg = <0x10>; + avdd-supply = <®_headphones_avdd>; + tvdd-supply = <&pm8916_l6>; + pdn-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphones_pdn_default>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index e5820235e0d5..0416c14895d6 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -10,6 +10,11 @@ Required properties: Optional properties: - reset-gpios: A GPIO specifier for the power down & reset pin - mute-gpios: A GPIO specifier for the soft mute pin +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply +- dsd-path: Select DSD input pins for ak4497 + 0: select #16, #17, #19 pins + 1: select #3, #4, #5 pins Example: diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt deleted file mode 100644 index 49a2e74fd9cb..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ /dev/null @@ -1,27 +0,0 @@ -AK4613 I2C transmitter - -This device supports I2C mode only. - -Required properties: - -- compatible : "asahi-kasei,ak4613" -- reg : The chip select number on the I2C bus - -Optional properties: -- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. -- asahi-kasei,in2-single-end rather than differential. -- asahi-kasei,out1-single-end -- asahi-kasei,out2-single-end -- asahi-kasei,out3-single-end -- asahi-kasei,out4-single-end -- asahi-kasei,out5-single-end -- asahi-kasei,out6-single-end - -Example: - -&i2c { - ak4613: ak4613@10 { - compatible = "asahi-kasei,ak4613"; - reg = <0x10>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml new file mode 100644 index 000000000000..ef4055ef0ccd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: codec@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt deleted file mode 100644 index 58e48ee97175..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4642.txt +++ /dev/null @@ -1,37 +0,0 @@ -AK4642 I2C transmitter - -This device supports I2C mode only. - -Required properties: - - - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" - - reg : The chip select number on the I2C bus - -Optional properties: - - - #clock-cells : common clock binding; shall be set to 0 - - clocks : common clock binding; MCKI clock - - clock-frequency : common clock binding; frequency of MCKO - - clock-output-names : common clock binding; MCKO clock name - -Example 1: - -&i2c { - ak4648: ak4648@12 { - compatible = "asahi-kasei,ak4642"; - reg = <0x12>; - }; -}; - -Example 2: - -&i2c { - ak4643: codec@12 { - compatible = "asahi-kasei,ak4643"; - reg = <0x12>; - #clock-cells = <0>; - clocks = <&audio_clock>; - clock-frequency = <12288000>; - clock-output-names = "ak4643_mcko"; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml new file mode 100644 index 000000000000..1e2caa29790e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4642 + - asahi-kasei,ak4643 + - asahi-kasei,ak4648 + + reg: + maxItems: 1 + + "#clock-cells": + const: 0 + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-frequency: + description: common clock binding; frequency of MCKO + + clock-output-names: + description: common clock name + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + #sound-dai-cells = <0>; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak5558.txt b/Documentation/devicetree/bindings/sound/ak5558.txt index 7d67ca6ced80..e28708db6686 100644 --- a/Documentation/devicetree/bindings/sound/ak5558.txt +++ b/Documentation/devicetree/bindings/sound/ak5558.txt @@ -4,12 +4,14 @@ This device supports I2C mode only. Required properties: -- compatible : "asahi-kasei,ak5558" +- compatible : "asahi-kasei,ak5558" or "asahi-kasei,ak5552". - reg : The I2C address of the device. Optional properties: - reset-gpios: A GPIO specifier for the power down & reset pin. +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply Example: diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..559aff13ae23 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,267 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + 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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + maxItems: 18 + items: + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + maxItems: 1 + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml index eb3992138eec..34f6ee9de392 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml @@ -8,7 +8,7 @@ title: Allwinner A10 I2S Controller Device Tree Bindings maintainers: - Chen-Yu Tsai <wens@csie.org> - - Maxime Ripard <maxime.ripard@bootlin.com> + - Maxime Ripard <mripard@kernel.org> properties: "#sound-dai-cells": @@ -20,10 +20,21 @@ properties: - const: allwinner,sun6i-a31-i2s - const: allwinner,sun8i-a83t-i2s - const: allwinner,sun8i-h3-i2s + - items: + - const: allwinner,sun8i-r40-i2s + - const: allwinner,sun8i-h3-i2s + - items: + - const: allwinner,sun8i-v3-i2s + - const: allwinner,sun8i-h3-i2s - const: allwinner,sun50i-a64-codec-i2s - items: - const: allwinner,sun50i-a64-i2s - const: allwinner,sun8i-h3-i2s + - const: allwinner,sun50i-h6-i2s + - const: allwinner,sun50i-r329-i2s + - items: + - const: allwinner,sun20i-d1-i2s + - const: allwinner,sun50i-r329-i2s reg: maxItems: 1 @@ -59,6 +70,8 @@ allOf: - allwinner,sun8i-a83t-i2s - allwinner,sun8i-h3-i2s - allwinner,sun50i-a64-codec-i2s + - allwinner,sun50i-h6-i2s + - allwinner,sun50i-r329-i2s then: required: @@ -68,13 +81,14 @@ allOf: properties: compatible: contains: - const: allwinner,sun8i-a83t-i2s + enum: + - allwinner,sun8i-a83t-i2s + - allwinner,sun8i-h3-i2s then: properties: dmas: minItems: 1 - maxItems: 2 items: - description: RX DMA Channel - description: TX DMA Channel diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml index e0284d8c3b63..444a432912bb 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -10,7 +10,7 @@ maintainers: - Chen-Yu Tsai <wens@csie.org> - Liam Girdwood <lgirdwood@gmail.com> - Mark Brown <broonie@kernel.org> - - Maxime Ripard <maxime.ripard@bootlin.com> + - Maxime Ripard <mripard@kernel.org> properties: "#sound-dai-cells": @@ -70,7 +70,9 @@ allOf: properties: compatible: contains: - const: allwinner,sun8i-h3-spdif + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif then: properties: diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml new file mode 100644 index 000000000000..3b764415c9ab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml @@ -0,0 +1,39 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-a64-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A64 Analog Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + compatible: + const: allwinner,sun50i-a64-codec-analog + + reg: + maxItems: 1 + + cpvdd-supply: + description: + Regulator for the headphone amplifier + +required: + - compatible + - reg + - cpvdd-supply + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun50i-a64-codec-analog"; + reg = <0x01f015c0 0x4>; + cpvdd-supply = <®_eldo1>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml new file mode 100644 index 000000000000..26eca21e1f0f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a23-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A23 Analog Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + compatible: + oneOf: + # FIXME: This is documented in the PRCM binding, but needs to be + # migrated here at some point + # - allwinner,sun8i-a23-codec-analog + - const: allwinner,sun8i-h3-codec-analog + - items: + - const: allwinner,sun8i-v3-codec-analog + - const: allwinner,sun8i-h3-codec-analog + - const: allwinner,sun8i-v3s-codec-analog + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun8i-h3-codec-analog"; + reg = <0x01f015c0 0x4>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml new file mode 100644 index 000000000000..19f111f40225 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a33-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A33 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + "#sound-dai-cells": + minimum: 0 + maximum: 1 + description: + A value of 0 is deprecated. When used, it only allows access to + the ADC/DAC and AIF1 (the CPU DAI), not the other two AIFs/DAIs. + + compatible: + oneOf: + - items: + - const: allwinner,sun50i-a64-codec + - const: allwinner,sun8i-a33-codec + - const: allwinner,sun8i-a33-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + audio-codec@1c22e00 { + #sound-dai-cells = <1>; + compatible = "allwinner,sun8i-a33-codec"; + reg = <0x01c22e00 0x400>; + interrupts = <0 29 4>; + clocks = <&ccu 47>, <&ccu 92>; + clock-names = "bus", "mod"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml new file mode 100644 index 000000000000..0705f91199a0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -0,0 +1,118 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,aiu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AIU audio output controller + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 2 + + compatible: + items: + - enum: + - amlogic,aiu-gxbb + - amlogic,aiu-gxl + - amlogic,aiu-meson8 + - amlogic,aiu-meson8b + - const: amlogic,aiu + + clocks: + items: + - description: AIU peripheral clock + - description: I2S peripheral clock + - description: I2S output clock + - description: I2S master clock + - description: I2S mixer clock + - description: SPDIF peripheral clock + - description: SPDIF output clock + - description: SPDIF master clock + - description: SPDIF master clock multiplexer + + clock-names: + items: + - const: pclk + - const: i2s_pclk + - const: i2s_aoclk + - const: i2s_mclk + - const: i2s_mixer + - const: spdif_pclk + - const: spdif_aoclk + - const: spdif_mclk + - const: spdif_mclk_sel + + interrupts: + items: + - description: I2S interrupt line + - description: SPDIF interrupt line + + interrupt-names: + items: + - const: i2s + - const: spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - clocks + - clock-names + - interrupts + - interrupt-names + - reg + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/gxbb-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/reset/amlogic,meson-gxbb-reset.h> + + aiu: audio-controller@5400 { + compatible = "amlogic,aiu-gxl", "amlogic,aiu"; + #sound-dai-cells = <2>; + reg = <0x5400 0x2ac>; + interrupts = <GIC_SPI 48 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 50 IRQ_TYPE_EDGE_RISING>; + interrupt-names = "i2s", "spdif"; + clocks = <&clkc CLKID_AIU_GLUE>, + <&clkc CLKID_I2S_OUT>, + <&clkc CLKID_AOCLK_GATE>, + <&clkc CLKID_CTS_AMCLK>, + <&clkc CLKID_MIXER_IFACE>, + <&clkc CLKID_IEC958>, + <&clkc CLKID_IEC958_GATE>, + <&clkc CLKID_CTS_MCLK_I958>, + <&clkc CLKID_CTS_I958>; + clock-names = "pclk", + "i2s_pclk", + "i2s_aoclk", + "i2s_mclk", + "i2s_mixer", + "spdif_pclk", + "spdif_aoclk", + "spdif_mclk", + "spdif_mclk_sel"; + resets = <&reset RESET_AIU>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt index 4330fc9dca6d..fa4545ed81ca 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt @@ -4,14 +4,22 @@ Required properties: - compatible: 'amlogic,axg-toddr' or 'amlogic,axg-toddr' or 'amlogic,g12a-frddr' or - 'amlogic,g12a-toddr' + 'amlogic,g12a-toddr' or + 'amlogic,sm1-frddr' or + 'amlogic,sm1-toddr' - reg: physical base address of the controller and length of memory mapped region. - interrupts: interrupt specifier for the fifo. - clocks: phandle to the fifo peripheral clock provided by the audio clock controller. -- resets: phandle to memory ARB line provided by the arb reset controller. +- resets: list of reset phandle, one for each entry reset-names. +- reset-names: should contain the following: + * "arb" : memory ARB line (required) + * "rst" : dedicated device reset line (optional) - #sound-dai-cells: must be 0. +- amlogic,fifo-depth: The size of the controller's fifo in bytes. This + is useful for determining certain configuration such + as the flush threshold of the fifo Example of FRDDR A on the A113 SoC: @@ -22,4 +30,5 @@ frddr_a: audio-controller@1c0 { interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; resets = <&arb AXG_ARB_FRDDR_A>; + fifo-depth = <512>; }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt index 73f473a9365f..716878107a24 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt @@ -2,7 +2,8 @@ Required properties: - compatible: 'amlogic,axg-pdm' or - 'amlogic,g12a-pdm' + 'amlogic,g12a-pdm' or + 'amlogic,sm1-pdm' - reg: physical base address of the controller and length of memory mapped region. - clocks: list of clock phandle, one for each entry clock-names. @@ -12,6 +13,9 @@ Required properties: * "sysclk" : dsp system clock - #sound-dai-cells: must be 0. +Optional property: +- resets: phandle to the dedicated reset line of the pdm input. + Example of PDM on the A113 SoC: pdm: audio-controller@ff632000 { diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt index 0b82504fa419..df92a4ecf288 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt @@ -2,7 +2,8 @@ Required properties: - compatible: 'amlogic,axg-spdifin' or - 'amlogic,g12a-spdifin' + 'amlogic,g12a-spdifin' or + 'amlogic,sm1-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: @@ -10,6 +11,9 @@ Required properties: * "refclk" : spdif input reference clock - #sound-dai-cells: must be 0. +Optional property: +- resets: phandle to the dedicated reset line of the spdif input. + Example on the A113 SoC: spdifin: audio-controller@400 { diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt index 826152730508..28381dd1f633 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt @@ -2,13 +2,17 @@ Required properties: - compatible: 'amlogic,axg-spdifout' or - 'amlogic,g12a-spdifout' + 'amlogic,g12a-spdifout' or + 'amlogic,sm1-spdifout' - clocks: list of clock phandle, one for each entry clock-names. - clock-names: should contain the following: * "pclk" : peripheral clock. * "mclk" : master clock - #sound-dai-cells: must be 0. +Optional property: +- resets: phandle to the dedicated reset line of the spdif output. + Example on the A113 SoC: spdifout: audio-controller@480 { diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt index 8835a43edfbb..5996c0cd89c2 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt @@ -4,7 +4,9 @@ Required properties: - compatible: 'amlogic,axg-tdmin' or 'amlogic,axg-tdmout' or 'amlogic,g12a-tdmin' or - 'amlogic,g12a-tdmout' + 'amlogic,g12a-tdmout' or + 'amlogic,sm1-tdmin' or + 'amlogic,sm1-tdmout - reg: physical base address of the controller and length of memory mapped region. - clocks: list of clock phandle, one for each entry clock-names. diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml new file mode 100644 index 000000000000..77469a45bb7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a Internal DAC Control Glue + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: amlogic,g12a-toacodec + - items: + - enum: + - amlogic,sm1-toacodec + - const: amlogic,g12a-toacodec + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + toacodec: audio-controller@740 { + compatible = "amlogic,g12a-toacodec"; + reg = <0x740 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOACODEC>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt index aa6c35570d31..4e8cd7eb7cec 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt @@ -1,10 +1,12 @@ * Amlogic HDMI Tx control glue Required properties: -- compatible: "amlogic,g12a-tohdmitx" +- compatible: "amlogic,g12a-tohdmitx" or + "amlogic,sm1-tohdmitx" - reg: physical base address of the controller and length of memory mapped region. - #sound-dai-cells: should be 1. +- resets: phandle to the dedicated reset line of the hdmitx glue. Example on the S905X2 SoC: @@ -12,6 +14,7 @@ tohdmitx: audio-controller@744 { compatible = "amlogic,g12a-tohdmitx"; reg = <0x0 0x744 0x0 0x4>; #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOHDMITX>; }; Example of an 'amlogic,axg-sound-card': diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml new file mode 100644 index 000000000000..b4b35edcb493 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -0,0 +1,114 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic GX sound card + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + items: + - const: amlogic,gx-sound-card + + audio-aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: list of auxiliary devices + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + description: |- + 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. + + audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + description: |- + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +patternProperties: + "^dai-link-[0-9]+$": + type: object + description: |- + dai-link child nodes: + Container for dai-link level properties and the CODEC sub-nodes. + There should be at least one (and probably more) subnode of this type + + properties: + dai-format: + $ref: /schemas/types.yaml#/definitions/string + enum: [ i2s, left-j, dsp_a ] + + mclk-fs: + $ref: /schemas/types.yaml#/definitions/uint32 + description: |- + Multiplication factor between the frame rate and master clock + rate + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + patternProperties: + "^codec-[0-9]+$": + type: object + description: |- + Codecs: + dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. dai-link representing + frontend links have no codec, therefore have no subnodes + + properties: + sound-dai: + maxItems: 1 + description: phandle of the codec DAI + + required: + - sound-dai + + required: + - sound-dai + +required: + - model + - dai-link-0 + +additionalProperties: false + +examples: + - | + sound { + compatible = "amlogic,gx-sound-card"; + model = "GXL-ACME-S905X-FOO"; + audio-aux-devs = <&>; + audio-routing = "I2S ENCODER I2S IN", "I2S FIFO Playback"; + + dai-link-0 { + sound-dai = <&i2s_fifo>; + }; + + dai-link-1 { + sound-dai = <&i2s_encoder>; + dai-format = "i2s"; + mclk-fs = <256>; + + codec-0 { + sound-dai = <&codec0>; + }; + + codec-1 { + sound-dai = <&codec1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml new file mode 100644 index 000000000000..580a3d040abc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic T9015 Internal Audio DAC + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 0 + + compatible: + items: + - const: amlogic,t9015 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: pclk + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + AVDD-supply: + description: + Analogue power supply. + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - resets + - AVDD-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/g12a-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-reset.h> + + acodec: audio-controller@32000 { + compatible = "amlogic,t9015"; + reg = <0x32000 0x14>; + #sound-dai-cells = <0>; + clocks = <&clkc CLKID_AUDIO_CODEC>; + clock-names = "pclk"; + resets = <&reset RESET_AUDIO_CODEC>; + AVDD-supply = <&vddao_1v8>; + }; diff --git a/Documentation/devicetree/bindings/sound/arm,pl041.yaml b/Documentation/devicetree/bindings/sound/arm,pl041.yaml new file mode 100644 index 000000000000..7896b8150cf0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arm,pl041.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/arm,pl041.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Arm Ltd. PrimeCell PL041 AACI sound interface + +maintainers: + - Andre Przywara <andre.przywara@arm.com> + +description: + The Arm PrimeCell Advanced Audio CODEC Interface (AACI) is an AMBA compliant + peripheral that provides communication with an audio CODEC using the AC-link + protocol. + +# We need a select here so we don't match all nodes with 'arm,primecell' +select: + properties: + compatible: + contains: + const: arm,pl041 + required: + - compatible + +properties: + compatible: + items: + - const: arm,pl041 + - const: arm,primecell + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + description: APB register access clock + + clock-names: + const: apb_pclk + +required: + - compatible + - reg + - interrupts + - clocks + +additionalProperties: false + +examples: + - | + audio-controller@40000 { + compatible = "arm,pl041", "arm,primecell"; + reg = <0x040000 0x1000>; + interrupts = <11>; + clocks = <&v2m_clk24mhz>; + clock-names = "apb_pclk"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt deleted file mode 100644 index 0e76946385ae..000000000000 --- a/Documentation/devicetree/bindings/sound/arndale.txt +++ /dev/null @@ -1,24 +0,0 @@ -Audio Binding for Arndale boards - -Required properties: -- compatible : Can be the following, - "samsung,arndale-rt5631" - -- samsung,audio-cpu: The phandle of the Samsung I2S controller -- samsung,audio-codec: The phandle of the audio codec - -Optional: -- samsung,model: The name of the sound-card - -Arndale Boards has many audio daughter cards, one of them is -rt5631/alc5631. Below example shows audio bindings for rt5631/ -alc5631 based codec. - -Example: - -sound { - compatible = "samsung,arndale-rt5631"; - - samsung,audio-cpu = <&i2s0> - samsung,audio-codec = <&rt5631>; -}; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt deleted file mode 100644 index 269682619a70..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ /dev/null @@ -1,337 +0,0 @@ -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 -- widgets -- routing -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- mclk-fs -- hp-det-gpio -- mic-det-gpio -- 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} - -Optional properties: -- pa-gpios: GPIO used to control external amplifier. - ------------------------ -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"; - ... - }; - }; - }; - }; - - ------------------------ -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-card.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml new file mode 100644 index 000000000000..109e55f9e597 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: /schemas/sound/audio-graph.yaml# + +properties: + compatible: + enum: + - audio-graph-card + - audio-graph-scu-card + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "audio-graph-card"; + + dais = <&cpu_port_a>; + }; + + cpu { + /* + * dai-controller own settings + */ + + port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + dai-format = "left_j"; + }; + }; + }; + + codec { + /* + * codec own settings + */ + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml new file mode 100644 index 000000000000..7416067c945e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-card2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card2 Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - audio-graph-card2 + links: + $ref: /schemas/types.yaml#/definitions/phandle-array + label: + maxItems: 1 + routing: + description: | + 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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + multi: + type: object + description: Multi-CPU/Codec node + dpcm: + type: object + description: DPCM node + codec2codec: + type: object + description: Codec to Codec node + +required: + - compatible + - links + +additionalProperties: false + +examples: + - | + sound { + compatible = "audio-graph-card2"; + + links = <&cpu_port>; + }; + + cpu { + compatible = "cpu-driver"; + + cpu_port: port { cpu_ep: endpoint { remote-endpoint = <&codec_ep>; }; }; + }; + + codec { + compatible = "codec-driver"; + + port { codec_ep: endpoint { remote-endpoint = <&cpu_ep>; }; }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml new file mode 100644 index 000000000000..5c368674d11a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-port.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card 'port' Node Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +allOf: + - $ref: /schemas/graph.yaml#/$defs/port-base + +properties: + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 +patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: /schemas/graph.yaml#/$defs/endpoint-base + properties: + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are + ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + frame-master: + description: Indicates dai-link frame master. + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + bitclock-master: + description: Indicates dai-link bit clock master + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + + dai-format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width-map: + description: Mapping of sample widths to slot widths. For hardware + that cannot support a fixed slot width or a slot width always + equal to sample width. A matrix of one or more 3-tuples. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - + description: Sample width in bits + minimum: 8 + maximum: 64 + - + description: Slot width in bits + minimum: 8 + maximum: 256 + - + description: Slot count + minimum: 1 + maximum: 64 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml new file mode 100644 index 000000000000..4b46794e5153 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + dais: + $ref: /schemas/types.yaml#/definitions/phandle-array + label: + maxItems: 1 + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + routing: + description: | + 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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + pa-gpios: + maxItems: 1 + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + +required: + - dais + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml new file mode 100644 index 000000000000..dce86dafe382 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw8738.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic AW8738 Audio Amplifier + +maintainers: + - Stephan Gerhold <stephan@gerhold.net> + +description: + The Awinic AW8738 is a simple audio amplifier with different operation modes + (set using one-wire pulse control). The mode configures the speaker-guard + function (primarily the power limit for the amplifier). + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: awinic,aw8738 + + mode-gpios: + description: + GPIO used for one-wire pulse control. The pin is typically called SHDN + (active-low), but this is misleading since it is actually more than + just a simple shutdown/enable control. + maxItems: 1 + + awinic,mode: + description: Operation mode (number of pulses for one-wire pulse control) + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 7 + + sound-name-prefix: true + +required: + - compatible + - mode-gpios + - awinic,mode + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + audio-amplifier { + compatible = "awinic,aw8738"; + mode-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + awinic,mode = <5>; + sound-name-prefix = "Speaker Amp"; + }; diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt new file mode 100644 index 000000000000..007f524b4d15 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt @@ -0,0 +1,29 @@ +Broadcom DSL/PON BCM63xx Audio I2S controller + +Required properties: +- compatible: Should be "brcm,bcm63xx-i2s". +- #address-cells: 32bit valued, 1 cell. +- #size-cells: 32bit valued, 0 cell. +- reg: Should contain audio registers location and length +- interrupts: Should contain the interrupt for the controller. +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: One of each entry matching the clocks phandles list: + - "i2sclk" (generated clock) Required. + - "i2sosc" (fixed 200MHz clock) Required. + +(1) : The generated clock is required only when any of TX and RX + works on Master Mode. +(2) : The fixed 200MHz clock is from internal chip and always on + +Example: + + i2s: bcm63xx-i2s { + #address-cells = <1>; + #size-cells = <0>; + compatible = "brcm,bcm63xx-i2s"; + reg = <0xFF802080 0xFF>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&i2sclk>, <&osc>; + clock-names = "i2sclk","i2sosc"; + }; diff --git a/Documentation/devicetree/bindings/sound/bt-sco.txt b/Documentation/devicetree/bindings/sound/bt-sco.txt deleted file mode 100644 index 641edf75e184..000000000000 --- a/Documentation/devicetree/bindings/sound/bt-sco.txt +++ /dev/null @@ -1,13 +0,0 @@ -Bluetooth-SCO audio CODEC - -This device support generic Bluetooth SCO link. - -Required properties: - - - compatible : "delta,dfbmcs320" or "linux,bt-sco" - -Example: - -codec: bt_sco { - compatible = "delta,dfbmcs320"; -}; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml new file mode 100644 index 000000000000..51d815d0c696 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -0,0 +1,195 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs35l41.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS35L41 Speaker Amplifier + +maintainers: + - david.rhodes@cirrus.com + +description: | + CS35L41 is a boosted mono Class D amplifier with DSP + speaker protection and equalization + +properties: + compatible: + enum: + - cirrus,cs35l40 + - cirrus,cs35l41 + + reg: + maxItems: 1 + + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + reset-gpios: + maxItems: 1 + + VA-supply: + description: voltage regulator phandle for the VA supply + + VP-supply: + description: voltage regulator phandle for the VP supply + + cirrus,boost-peak-milliamp: + description: + Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600 mA and goes to a maximum of 4500 mA with increments + of 50 mA. See section 4.3.6 of the datasheet for details. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 1600 + maximum: 4500 + default: 4500 + + cirrus,boost-ind-nanohenry: + description: + Boost inductor value, expressed in nH. Valid + values include 1000, 1200, 1500 and 2200. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 1000 + maximum: 2200 + + cirrus,boost-cap-microfarad: + description: + Total equivalent boost capacitance on the VBST + and VAMP pins, derated at 11 volts DC. The value must be rounded to the + nearest integer and expressed in uF. + $ref: "/schemas/types.yaml#/definitions/uint32" + + cirrus,asp-sdout-hiz: + description: + Audio serial port SDOUT Hi-Z control. Sets the Hi-Z + configuration for SDOUT pin of amplifier. + 0 = Logic 0 during unused slots, and while all transmit channels disabled + 1 = Hi-Z during unused slots but logic 0 while all transmit channels disabled + 2 = (Default) Logic 0 during unused slots, but Hi-Z while all transmit channels disabled + 3 = Hi-Z during unused slots and while all transmit channels disabled + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + default: 2 + + cirrus,boost-type: + description: + Configures the type of Boost being used. + Internal boost requires boost-peak-milliamp, boost-ind-nanohenry and + boost-cap-microfarad. + External Boost must have GPIO1 as GPIO output. GPIO1 will be set high to + enable boost voltage. + 0 = Internal Boost + 1 = External Boost + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 1 + + cirrus,gpio1-polarity-invert: + description: + Boolean which specifies whether the GPIO1 + level is inverted. If this property is not present the level is not inverted. + type: boolean + + cirrus,gpio1-output-enable: + description: + Boolean which specifies whether the GPIO1 pin + is configured as an output. If this property is not present the + pin will be configured as an input. + type: boolean + + cirrus,gpio1-src-select: + description: + Configures the function of the GPIO1 pin. + Note that the options are different from the GPIO2 pin + 0 = High Impedance (Default) + 1 = GPIO + 2 = Sync + 3 = MCLK input + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + + cirrus,gpio2-polarity-invert: + description: + Boolean which specifies whether the GPIO2 + level is inverted. If this property is not present the level is not inverted. + type: boolean + + cirrus,gpio2-output-enable: + description: + Boolean which specifies whether the GPIO2 pin + is configured as an output. If this property is not present the + pin will be configured as an input. + type: boolean + + cirrus,gpio2-src-select: + description: + Configures the function of the GPIO2 pin. + Note that the options are different from the GPIO1 pin. + 0 = High Impedance (Default) + 1 = GPIO + 2 = Open Drain INTB + 3 = MCLK input + 4 = Push-pull INTB (active low) + 5 = Push-pull INT (active high) + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 5 + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - if: + properties: + cirrus,boost-type: + const: 0 + then: + required: + - cirrus,boost-peak-milliamp + - cirrus,boost-ind-nanohenry + - cirrus,boost-cap-microfarad + else: + if: + properties: + cirrus,boost-type: + const: 1 + then: + required: + - cirrus,gpio1-output-enable + - cirrus,gpio1-src-select + properties: + cirrus,boost-peak-milliamp: false + cirrus,boost-ind-nanohenry: false + cirrus,boost-cap-microfarad: false + cirrus,gpio1-src-select: + enum: [1] + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l41: cs35l41@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + + cirrus,boost-type = <0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml new file mode 100644 index 000000000000..184a1344ea76 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs35l45.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS35L45 Speaker Amplifier + +maintainers: + - Ricardo Rivera-Matos <rriveram@opensource.cirrus.com> + - Richard Fitzgerald <rf@opensource.cirrus.com> + +description: | + CS35L45 is a Boosted Mono Class D Amplifier with DSP + Speaker Protection and Adaptive Battery Management. + +properties: + compatible: + enum: + - cirrus,cs35l45 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + reset-gpios: + maxItems: 1 + + vdd-a-supply: + description: voltage regulator phandle for the VDD_A supply + + vdd-batt-supply: + description: voltage regulator phandle for the VDD_BATT supply + + spi-max-frequency: + maximum: 5000000 + + cirrus,asp-sdout-hiz-ctrl: + description: + Audio serial port SDOUT Hi-Z control. Sets the Hi-Z + configuration for SDOUT pin of amplifier. Logical OR of + CS35L45_ASP_TX_HIZ_xxx values. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + default: 2 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs35l45.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l45: cs35l45@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l45"; + reg = <2>; + spi-max-frequency = <5000000>; + vdd-a-supply = <&dummy_vreg>; + vdd-batt-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + cirrus,asp-sdout-hiz-ctrl = <(CS35L45_ASP_TX_HIZ_UNUSED | + CS35L45_ASP_TX_HIZ_DISABLED)>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml new file mode 100644 index 000000000000..156560b2a980 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs4234.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic cs4234 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: + The CS4234 is a highly versatile CODEC that combines 4 channels of + high performance analog to digital conversion, 4 channels of high + performance digital to analog conversion for audio, and 1 channel of + digital to analog conversion to provide a nondelayed audio reference + signal to an external Class H tracking power supply. If not used to + drive a tracking power supply, the 5th DAC can instead be used as a + standard audio grade DAC, with performance specifications identical + to that of the 4 DACs in the audio path. Additionally, the CS4234 + includes tunable group delay for each of the 4 audio DAC paths to + provide lead time for the external switch-mode power supply, and a + nondelayed path into the DAC outputs for input signals requiring a + low-latency path to the outputs. + +properties: + compatible: + enum: + - cirrus,cs4234 + + reg: + description: + The 7-bit I2C address depends on the state of the ADx pins, in + binary given by [0 0 1 0 AD2 AD1 AD0 0]. + items: + minimum: 0x10 + maximum: 0x17 + + VA-supply: + description: + Analogue power supply. + + VL-supply: + description: + Interface power supply. + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - VA-supply + - VL-supply + +additionalProperties: false + +examples: + - | + i2c@e0004000 { + #address-cells = <1>; + #size-cells = <0>; + reg = <0xe0004000 0x1000>; + + cs4234: codec@11 { + compatible = "cirrus,cs4234"; + reg = <0x11>; + + VA-supply = <&vdd3v3>; + VL-supply = <&vdd3v3>; + + reset-gpios = <&gpio 0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml new file mode 100644 index 000000000000..31800f70e9d9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml @@ -0,0 +1,225 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l42.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS42L42 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: + The CS42L42 is a low-power audio codec designed for portable applications. + It provides a high-dynamic range, stereo DAC for audio playback and a mono + high-dynamic-range ADC for audio capture. There is an integrated headset + detection block. + +properties: + compatible: + enum: + - cirrus,cs42l42 + + reg: + description: + The I2C address of the CS42L42. + maxItems: 1 + + VP-supply: + description: + VP power supply. + + VCP-supply: + description: + Charge pump power supply. + + VD_FILT-supply: + description: + FILT+ power supply. + + VL-supply: + description: + Logic power supply. + + VA-supply: + description: + Analog power supply. + + reset-gpios: + description: + This pin will be asserted and then deasserted to reset the + CS42L42 before communication starts. + maxItems: 1 + + interrupts: + description: + Interrupt for CS42L42 IRQ line. + maxItems: 1 + + cirrus,ts-inv: + description: | + Sets the behaviour of the jack plug detect switch. + + 0 - (Default) Shorted to tip when unplugged, open when plugged. + This is "inverted tip sense (ITS)" in the datasheet. + + 1 - Open when unplugged, shorted to tip when plugged. + This is "normal tip sense (TS)" in the datasheet. + + The CS42L42_TS_INV_* defines are available for this. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 1 + + cirrus,ts-dbnc-rise: + description: | + Debounce the rising edge of TIP_SENSE_PLUG. With no + debounce, the tip sense pin might be noisy on a plug event. + + 0 - 0ms + 1 - 125ms + 2 - 250ms + 3 - 500ms + 4 - 750ms + 5 - 1s (Default) + 6 - 1.25s + 7 - 1.5s + + The CS42L42_TS_DBNCE_* defines are available for this. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 7 + + cirrus,ts-dbnc-fall: + description: | + Debounce the falling edge of TIP_SENSE_UNPLUG. With no + debounce, the tip sense pin might be noisy on an unplug event. + + 0 - 0ms + 1 - 125ms + 2 - 250ms + 3 - 500ms + 4 - 750ms + 5 - 1s (Default) + 6 - 1.25s + 7 - 1.5s + + The CS42L42_TS_DBNCE_* defines are available for this. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 7 + + cirrus,btn-det-init-dbnce: + description: | + This sets how long to wait after enabling button detection + interrupts before servicing button interrupts, to allow the + HS bias time to settle. Value is in milliseconds. + There may be erroneous button interrupts if this debounce time + is too short. + + 0ms - 200ms, + Default = 100ms + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 200 + + cirrus,btn-det-event-dbnce: + description: | + This sets how long to wait after receiving a button press + interrupt before processing it. Allows time for the button + press to make a clean connection with the bias resistors. + Value is in milliseconds. + + 0ms - 20ms, + Default = 10ms + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 20 + + cirrus,bias-lvls: + description: | + For a level-detect headset button scheme, each button will bias + the mic pin to a certain voltage. To determine which button was + pressed, the voltage is compared to sequential, decreasing + voltages, until the compared voltage < bias voltage. + For different hardware setups, a designer might want to tweak this. + This is an array of descending values for the comparator voltage, + given as percent of the HSBIAS voltage. + + Array of 4 values, each 0-63 + < x1 x2 x3 x4 > + Default = < 15 8 4 1 > + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + items: + minimum: 0 + maximum: 63 + + cirrus,hs-bias-ramp-rate: + description: | + If present this sets the rate that the HS bias should rise and fall. + The actual rise and fall times depend on external hardware (the + datasheet gives several rise and fall time examples). + + 0 - Fast rise time; slow, load-dependent fall time + 1 - Fast + 2 - Slow (default) + 3 - Slowest + + The CS42L42_HSBIAS_RAMP_* defines are available for this. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + + cirrus,hs-bias-sense-disable: + description: | + If present the HSBIAS sense is disabled. Configures HSBIAS output + current sense through the external 2.21-k resistor. HSBIAS_SENSE + is a hardware feature to reduce the potential pop noise when the + headset plug is removed slowly. But on some platforms ESD voltage + will affect it causing plug detection to fail, especially with CTIA + headset type. For different hardware setups, a designer might want + to tweak default behavior. + type: boolean + +required: + - compatible + - reg + - VP-supply + - VCP-supply + - VD_FILT-supply + - VL-supply + - VA-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs42l42.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l42: cs42l42@48 { + compatible = "cirrus,cs42l42"; + reg = <0x48>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD_FILT-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + + reset-gpios = <&axi_gpio_0 1 0>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + + cirrus,ts-inv = <CS42L42_TS_INV_DIS>; + cirrus,ts-dbnc-rise = <CS42L42_TS_DBNCE_1000>; + cirrus,ts-dbnc-fall = <CS42L42_TS_DBNCE_0>; + cirrus,btn-det-init-dbnce = <100>; + cirrus,btn-det-event-dbnce = <10>; + cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>; + cirrus,hs-bias-ramp-rate = <CS42L42_HSBIAS_RAMP_SLOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml new file mode 100644 index 000000000000..963a871e74da --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l51.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: CS42L51 audio codec DT bindings + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +properties: + compatible: + const: cirrus,cs42l51 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: MCLK + + reset-gpios: + maxItems: 1 + + VL-supply: + description: phandle to voltage regulator of digital interface section + + VD-supply: + description: phandle to voltage regulator of digital internal section + + VA-supply: + description: phandle to voltage regulator of analog internal section + + VAHP-supply: + description: phandle to voltage regulator of headphone + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l51@4a { + compatible = "cirrus,cs42l51"; + reg = <0x4a>; + #sound-dai-cells = <0>; + clocks = <&mclk_prov>; + clock-names = "MCLK"; + VL-supply = <®_audio>; + VD-supply = <®_audio>; + VA-supply = <®_audio>; + VAHP-supply = <®_audio>; + reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt deleted file mode 100644 index 41ae2699f07a..000000000000 --- a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt +++ /dev/null @@ -1,39 +0,0 @@ -Cirrus Logic Lochnagar Audio Development Board - -Lochnagar is an evaluation and development board for Cirrus Logic -Smart CODEC and Amp devices. It allows the connection of most Cirrus -Logic devices on mini-cards, as well as allowing connection of -various application processor systems to provide a full evaluation -platform. Audio system topology, clocking and power can all be -controlled through the Lochnagar, allowing the device under test -to be used in a variety of possible use cases. - -This binding document describes the binding for the audio portion -of the driver. - -This binding must be part of the Lochnagar MFD binding: - [4] ../mfd/cirrus,lochnagar.txt - -Required properties: - - - compatible : One of the following strings: - "cirrus,lochnagar2-soundcard" - - - #sound-dai-cells : Must be set to 1. - - - clocks : Contains an entry for each entry in clock-names. - - clock-names : Must include the following clocks: - "mclk" Master clock source for the sound card, should normally - be set to LOCHNAGAR_SOUNDCARD_MCLK provided by the Lochnagar - clock driver. - -Example: - -lochnagar-sc { - compatible = "cirrus,lochnagar2-soundcard"; - - #sound-dai-cells = <1>; - - clocks = <&lochnagar_clk LOCHNAGAR_SOUNDCARD_MCLK>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml new file mode 100644 index 000000000000..cea612d3d4a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,lochnagar.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic Lochnagar Audio Development Board + +maintainers: + - patches@opensource.cirrus.com + +description: | + Lochnagar is an evaluation and development board for Cirrus Logic + Smart CODEC and Amp devices. It allows the connection of most Cirrus + Logic devices on mini-cards, as well as allowing connection of various + application processor systems to provide a full evaluation platform. + Audio system topology, clocking and power can all be controlled through + the Lochnagar, allowing the device under test to be used in a variety of + possible use cases. + + This binding document describes the binding for the audio portion of the + driver. + + This binding must be part of the Lochnagar MFD binding: + [1] ../mfd/cirrus,lochnagar.yaml + +properties: + compatible: + enum: + - cirrus,lochnagar2-soundcard + + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + clocks: + description: + Master clock source for the sound card, should normally be set to + LOCHNAGAR_SOUNDCARD_MCLK provided by the Lochnagar clock driver. + maxItems: 1 + + clock-names: + const: mclk + +required: + - compatible + - '#sound-dai-cells' + - clocks + - clock-names + +additionalProperties: false diff --git a/Documentation/devicetree/bindings/sound/cirrus,madera.yaml b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml new file mode 100644 index 000000000000..23138ddcb62d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,madera.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic Madera class audio CODECs + +maintainers: + - patches@opensource.cirrus.com + +description: | + This describes audio configuration bindings for these codecs. + + See also the core bindings for the parent MFD driver: + + Documentation/devicetree/bindings/mfd/cirrus,madera.yaml + + and defines for values used in these bindings: + + include/dt-bindings/sound/madera.h + + The properties are all contained in the parent MFD node. + +properties: + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + cirrus,inmode: + description: + A list of input mode settings for each input. A maximum + of 24 cells, with four cells per input in the order INnAL, + INnAR INnBL INnBR. For non-muxed inputs the first two cells + for that input set the mode for the left and right channel + and the second two cells must be 0. For muxed inputs the + first two cells for that input set the mode of the left and + right A inputs and the second two cells set the mode of the + left and right B inputs. Valid mode values are one of the + MADERA_INMODE_xxx. If the array is shorter than the number + of inputs the unspecified inputs default to MADERA_INMODE_DIFF. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 24 + items: + minimum: 0 + maximum: 1 + default: 0 + + cirrus,out-mono: + description: + Mono bit for each output, maximum of six cells if the array + is shorter outputs will be set to stereo. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 6 + items: + minimum: 0 + maximum: 1 + default: 0 + + cirrus,dmic-ref: + description: | + Indicates how the MICBIAS pins have been externally connected + to DMICs on each input, one cell per input. + + <IN1 IN2 IN3 ...> + + A value of 0 indicates MICVDD and is the default, + other values depend on the codec: For CS47L35 one of the + CS47L35_DMIC_REF_xxx values For all other codecs one of + the MADERA_DMIC_REF_xxx values Also see the datasheet for a + description of the INn_DMIC_SUP field. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 6 + items: + minimum: 0 + maximum: 3 + default: 0 + + cirrus,max-channels-clocked: + description: + Maximum number of channels that I2S clocks will be generated + for. Useful when clock master for systems where the I2S bus + has multiple data lines. One cell for each AIF, use a value + of zero for AIFs that should be handled normally. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + default: 0 + + cirrus,pdm-fmt: + description: + PDM speaker data format, must contain 2 cells (OUT5 and + OUT6). See the PDM_SPKn_FMT field in the datasheet for a + description of this value. The second cell is ignored for + codecs that do not have OUT6. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + cirrus,pdm-mute: + description: | + PDM mute format, must contain 2 cells (OUT5 and OUT6). See the + PDM_SPKn_CTRL_1 register in the datasheet for a description + of this value. The second cell is ignored for codecs that + do not have OUT6. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/cs42l42.txt b/Documentation/devicetree/bindings/sound/cs42l42.txt deleted file mode 100644 index 7dfaa2ab906f..000000000000 --- a/Documentation/devicetree/bindings/sound/cs42l42.txt +++ /dev/null @@ -1,107 +0,0 @@ -CS42L42 audio CODEC - -Required properties: - - - compatible : "cirrus,cs42l42" - - - reg : the I2C address of the device for I2C. - - - VP-supply, VCP-supply, VD_FILT-supply, VL-supply, VA-supply : - power supplies for the device, as covered in - Documentation/devicetree/bindings/regulator/regulator.txt. - -Optional properties: - - - reset-gpios : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the codec starts. - - - interrupts : IRQ line info CS42L42. - (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt - for further information relating to interrupt properties) - - - cirrus,ts-inv : Boolean property. For jacks that invert the tip sense - polarity. Normal jacks will short tip sense pin to HS1 when headphones are - plugged in and leave tip sense floating when not plugged in. Inverting jacks - short tip sense when unplugged and float when plugged in. - - 0 = (Default) Non-inverted - 1 = Inverted - - - cirrus,ts-dbnc-rise : Debounce the rising edge of TIP_SENSE_PLUG. With no - debounce, the tip sense pin might be noisy on a plug event. - - 0 - 0ms, - 1 - 125ms, - 2 - 250ms, - 3 - 500ms, - 4 - 750ms, - 5 - (Default) 1s, - 6 - 1.25s, - 7 - 1.5s, - - - cirrus,ts-dbnc-fall : Debounce the falling edge of TIP_SENSE_UNPLUG. - With no debounce, the tip sense pin might be noisy on an unplug event. - - 0 - 0ms, - 1 - 125ms, - 2 - 250ms, - 3 - 500ms, - 4 - 750ms, - 5 - (Default) 1s, - 6 - 1.25s, - 7 - 1.5s, - - - cirrus,btn-det-init-dbnce : This sets how long the driver sleeps after - enabling button detection interrupts. After auto-detection and before - servicing button interrupts, the HS bias needs time to settle. If you - don't wait, there is possibility for erroneous button interrupt. - - 0ms - 200ms, - Default = 100ms - - - cirrus,btn-det-event-dbnce : This sets how long the driver delays after - receiving a button press interrupt. With level detect interrupts, you want - to wait a small amount of time to make sure the button press is making a - clean connection with the bias resistors. - - 0ms - 20ms, - Default = 10ms - - - cirrus,bias-lvls : For a level-detect headset button scheme, each button - will bias the mic pin to a certain voltage. To determine which button was - pressed, the driver will compare this biased voltage to sequential, - decreasing voltages and will stop when a comparator is tripped, - indicating a comparator voltage < bias voltage. This value represents a - percentage of the internally generated HS bias voltage. For different - hardware setups, a designer might want to tweak this. This is an array of - descending values for the comparator voltage. - - Array of 4 values - Each 0-63 - < x1 x2 x3 x4 > - Default = < 15 8 4 1> - - -Example: - -cs42l42: cs42l42@48 { - compatible = "cirrus,cs42l42"; - reg = <0x48>; - VA-supply = <&dummy_vreg>; - VP-supply = <&dummy_vreg>; - VCP-supply = <&dummy_vreg>; - VD_FILT-supply = <&dummy_vreg>; - VL-supply = <&dummy_vreg>; - - reset-gpios = <&axi_gpio_0 1 0>; - interrupt-parent = <&gpio0>; - interrupts = <55 8> - - cirrus,ts-inv = <0x00>; - cirrus,ts-dbnc-rise = <0x05>; - cirrus,ts-dbnc-fall = <0x00>; - cirrus,btn-det-init-dbnce = <100>; - cirrus,btn-det-event-dbnce = <10>; - cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>; - cirrus,hs-bias-ramp-rate = <0x02>; -}; diff --git a/Documentation/devicetree/bindings/sound/cs42l51.txt b/Documentation/devicetree/bindings/sound/cs42l51.txt deleted file mode 100644 index acbd68ddd2cb..000000000000 --- a/Documentation/devicetree/bindings/sound/cs42l51.txt +++ /dev/null @@ -1,33 +0,0 @@ -CS42L51 audio CODEC - -Required properties: - - - compatible : "cirrus,cs42l51" - - - reg : the I2C address of the device for I2C. - -Optional properties: - - VL-supply, VD-supply, VA-supply, VAHP-supply: power supplies for the device, - as covered in Documentation/devicetree/bindings/regulator/regulator.txt. - - - reset-gpios : GPIO specification for the reset pin. If specified, it will be - deasserted before starting the communication with the codec. - - - clocks : a list of phandles + clock-specifiers, one for each entry in - clock-names - - - clock-names : must contain "MCLK" - -Example: - -cs42l51: cs42l51@4a { - compatible = "cirrus,cs42l51"; - reg = <0x4a>; - clocks = <&mclk_prov>; - clock-names = "MCLK"; - VL-supply = <®_audio>; - VD-supply = <®_audio>; - VA-supply = <®_audio>; - VAHP-supply = <®_audio>; - reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; -}; diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt index 58902802d56c..94584c96c4ae 100644 --- a/Documentation/devicetree/bindings/sound/da7213.txt +++ b/Documentation/devicetree/bindings/sound/da7213.txt @@ -1,9 +1,9 @@ -Dialog Semiconductor DA7213 Audio Codec bindings +Dialog Semiconductor DA7212/DA7213 Audio Codec bindings ====== Required properties: -- compatible : Should be "dlg,da7213" +- compatible : Should be "dlg,da7212" or "dlg,da7213" - reg: Specifies the I2C slave address Optional properties: @@ -21,6 +21,10 @@ Optional properties: - dlg,dmic-clkrate : DMIC clock frequency (Hz). [<1500000>, <3000000>] + - VDDA-supply : Regulator phandle for Analogue power supply + - VDDMIC-supply : Regulator phandle for Mic Bias + - VDDIO-supply : Regulator phandle for I/O power supply + ====== Example: diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index c483dcec01f8..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,85 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles 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. The dma - identifiers must be "rx" and "tx". - -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 -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -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 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..f46c66bc6b2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary@ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: 0 - I2S or 1 - DIT operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number of channels over one serializer + the property is ignored in DIT mode + minimum: 2 + maximum: 32 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures WFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures RFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX interrupt + - description: RX interrupt + - items: + - description: common/combined interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +allOf: + - if: + properties: + opmode: + enum: + - 0 + + then: + required: + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000000..3b752bba748b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8316 audio CODEC + +maintainers: + - Daniel Drake <drake@endlessm.com> + - Katsuhiro Suzuki <katsuhiro@katsuster.net> + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..998b4c8a7f78 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,12 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-asrc", + "fsl,imx53-asrc", + "fsl,imx8qm-asrc", + "fsl,imx8qxp-asrc", - reg : Offset and length of the register set for the device. @@ -35,12 +40,21 @@ Required properties: - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + - fsl,asrc-clk-map : Defines clock map used in driver. which is required + by imx8qm/imx8qxp platform + <0> - select the map for asrc0 in imx8qm/imx8qxp + <1> - select the map for asrc1 in imx8qm/imx8qxp + Optional properties: - big-endian : If this property is absent, the little endian mode will be in use as default. Otherwise, the big endian mode will be in use for all the device registers. + - fsl,asrc-format : Defines a mutual sample format used by DPCM Back + Ends, which can replace the fsl,asrc-width. + The value is 2 (S16_LE), or 6 (S24_LE). + Example: asrc: asrc@2034000 { diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..aa4be7170717 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml new file mode 100644 index 000000000000..bdde68a1059c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,easrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Asynchronous Sample Rate Converter (ASRC) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + $nodename: + pattern: "^easrc@.*" + + compatible: + const: fsl,imx8mn-easrc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: mem + + dmas: + maxItems: 8 + + dma-names: + items: + - const: ctx0_rx + - const: ctx0_tx + - const: ctx1_rx + - const: ctx1_tx + - const: ctx2_rx + - const: ctx2_tx + - const: ctx3_rx + - const: ctx3_tx + + firmware-name: + $ref: /schemas/types.yaml#/definitions/string + const: imx/easrc/easrc-imx8mn.bin + description: The coefficient table for the filters + + fsl,asrc-rate: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 8000 + maximum: 192000 + description: Defines a mutual sample rate used by DPCM Back Ends + + fsl,asrc-format: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [2, 6, 10, 32, 36] + default: 2 + description: + Defines a mutual sample format used by DPCM Back Ends + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - firmware-name + - fsl,asrc-rate + - fsl,asrc-format + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + easrc: easrc@300c0000 { + compatible = "fsl,imx8mn-easrc"; + reg = <0x300c0000 0x10000>; + interrupts = <0x0 122 0x4>; + clocks = <&clk IMX8MN_CLK_ASRC_ROOT>; + clock-names = "mem"; + dmas = <&sdma2 16 23 0> , <&sdma2 17 23 0>, + <&sdma2 18 23 0> , <&sdma2 19 23 0>, + <&sdma2 20 23 0> , <&sdma2 21 23 0>, + <&sdma2 22 23 0> , <&sdma2 23 23 0>; + dma-names = "ctx0_rx", "ctx0_tx", + "ctx1_rx", "ctx1_tx", + "ctx2_rx", "ctx2_tx", + "ctx3_rx", "ctx3_tx"; + firmware-name = "imx/easrc/easrc-imx8mn.bin"; + fsl,asrc-rate = <8000>; + fsl,asrc-format = <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt index 5b9914367610..0a2480aeecf0 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -7,8 +7,12 @@ other DSPs. It has up to six transmitters and four receivers. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-esai" or - "fsl,vf610-esai" + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-esai", + "fsl,vf610-esai", + "fsl,imx6ull-esai", + "fsl,imx8qm-esai", - reg : Offset and length of the register set for the device. diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.txt b/Documentation/devicetree/bindings/sound/fsl,micfil.txt index 53e227b15277..1ea05d4996c7 100644 --- a/Documentation/devicetree/bindings/sound/fsl,micfil.txt +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.txt @@ -6,6 +6,7 @@ microphone bitstream in a configurable output sampling rate. Required properties: - compatible : Compatible list, contains "fsl,imx8mm-micfil" + or "fsl,imx8mp-micfil" - reg : Offset and length of the register set for the device. diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..40353fc30255 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,36 @@ +fsl,mqs audio CODEC + +Required properties: + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : "mclk" - must required. + "core" - required if compatible is "fsl,imx8qm-mqs", it + is for register access. + - gpr : A phandle of General Purpose Registers in IOMUX Controller. + Required if compatible is "fsl,imx6sx-mqs". + +Required if compatible is "fsl,imx8qm-mqs": + - power-domains: A phandle of PM domain provider node. + - reg: Offset and length of the register set for the device. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; + +mqs: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&clk IMX8QM_AUD_MQS_IPG>, + <&clk IMX8QM_AUD_MQS_HMCLK>; + clock-names = "core", "mclk"; + power-domains = <&pd_mqs0>; + status = "disabled"; +}; diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml new file mode 100644 index 000000000000..d370c98a62c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio RPMSG CPU DAI Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + fsl_rpmsg is a virtual audio device. Mapping to real hardware devices + are SAI, DMA controlled by Cortex M core. What we see from Linux + side is a device which provides audio service by rpmsg channel. + +properties: + compatible: + enum: + - fsl,imx7ulp-rpmsg-audio + - fsl,imx8mn-rpmsg-audio + - fsl,imx8mm-rpmsg-audio + - fsl,imx8mp-rpmsg-audio + - fsl,imx8ulp-rpmsg-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + clocks: + items: + - description: Peripheral clock for register access + - description: Master clock + - description: DMA clock for DMA register access + - description: Parent clock for multiple of 8kHz sample rates + - description: Parent clock for multiple of 11kHz sample rates + + clock-names: + items: + - const: ipg + - const: mclk + - const: dma + - const: pll8k + - const: pll11k + + power-domains: + description: + List of phandle and PM domain specifier as documented in + Documentation/devicetree/bindings/power/power_domain.txt + maxItems: 1 + + memory-region: + maxItems: 1 + description: + phandle to a node describing reserved memory (System RAM memory) + The M core can't access all the DDR memory space on some platform, + So reserved a specific memory for dma buffer which M core can + access. + (see bindings/reserved-memory/reserved-memory.txt) + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to a node of audio codec + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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. + + fsl,enable-lpa: + $ref: /schemas/types.yaml#/definitions/flag + description: enable low power audio path. + + fsl,rpmsg-out: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, the transmitting function + will be enabled. + + fsl,rpmsg-in: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, the receiving function + will be enabled. + +required: + - compatible + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + rpmsg_audio: rpmsg_audio { + compatible = "fsl,imx8mn-rpmsg-audio"; + model = "wm8524-audio"; + fsl,enable-lpa; + fsl,rpmsg-out; + clocks = <&clk IMX8MN_CLK_SAI3_IPG>, + <&clk IMX8MN_CLK_SAI3_ROOT>, + <&clk IMX8MN_CLK_SDMA3_ROOT>, + <&clk IMX8MN_AUDIO_PLL1_OUT>, + <&clk IMX8MN_AUDIO_PLL2_OUT>; + clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt deleted file mode 100644 index 8b324f82a782..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ /dev/null @@ -1,64 +0,0 @@ -Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Freescale S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - - - compatible : Compatible list, must contain "fsl,imx35-spdif". - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock of spdif controller. - "rxtx<0-7>" Clock source list for tx and rx clock. - This clock list should be identical to the source - list connecting to the spdif clock mux in "SPDIF - Transceiver Clock Diagram" of SoC reference manual. - It can also be referred to TxClk_Source bit of - register SPDIF_STC. - "spba" The spba clock is required when SPDIF is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -spdif: spdif@2004000 { - compatible = "fsl,imx35-spdif"; - reg = <0x02004000 0x4000>; - interrupts = <0 52 0x04>; - dmas = <&sdma 14 18 0>, - <&sdma 15 18 0>; - dma-names = "rx", "tx"; - - clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, - <&clks 0>, <&clks 118>, - <&clks 62>, <&clks 139>, - <&clks 0>; - clock-names = "core", "rxtx0", - "rxtx1", "rxtx2", - "rxtx3", "rxtx4", - "rxtx5", "rxtx6", - "rxtx7"; - - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 000000000000..f226ec13167a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + The Freescale S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via an coaxial cable or + a fibre cable. + +properties: + compatible: + enum: + - fsl,imx35-spdif + - fsl,vf610-spdif + - fsl,imx6sx-spdif + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8mq-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif + - fsl,imx8ulp-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + clocks: + items: + - description: The core clock of spdif controller. + - description: Clock for tx0 and rx0. + - description: Clock for tx1 and rx1. + - description: Clock for tx2 and rx2. + - description: Clock for tx3 and rx3. + - description: Clock for tx4 and rx4. + - description: Clock for tx5 and rx5. + - description: Clock for tx6 and rx6. + - description: Clock for tx7 and rx7. + - description: The spba clock is required when SPDIF is placed as a bus + slave of the Shared Peripheral Bus and when two or more bus masters + (CPU, DMA or DSP) try to access it. This property is optional depending + on the SoC design. + minItems: 9 + + clock-names: + items: + - const: core + - const: rxtx0 + - const: rxtx1 + - const: rxtx2 + - const: rxtx3 + - const: rxtx4 + - const: rxtx5 + - const: rxtx6 + - const: rxtx7 + - const: spba + minItems: 9 + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml new file mode 100644 index 000000000000..223b8ea693dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml @@ -0,0 +1,104 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,xcvr.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Transceiver (XCVR) Controller + +maintainers: + - Viorel Suman <viorel.suman@nxp.com> + +description: | + NXP XCVR (Audio Transceiver) is a on-chip functional module + that allows CPU to receive and transmit digital audio via + HDMI2.1 eARC, HDMI1.4 ARC and SPDIF. + +properties: + $nodename: + pattern: "^xcvr@.*" + + compatible: + enum: + - fsl,imx8mp-xcvr + + reg: + items: + - description: 20K RAM for code and data + - description: registers space + - description: RX FIFO address + - description: TX FIFO address + + reg-names: + items: + - const: ram + - const: regs + - const: rxfifo + - const: txfifo + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + - description: PHY clock + - description: SPBA clock + - description: PLL clock + + clock-names: + items: + - const: ipg + - const: phy + - const: spba + - const: pll_ipg + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + +required: + - compatible + - reg + - reg-names + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + #include <dt-bindings/reset/imx8mp-reset.h> + + xcvr: xcvr@30cc0000 { + compatible = "fsl,imx8mp-xcvr"; + reg = <0x30cc0000 0x800>, + <0x30cc0800 0x400>, + <0x30cc0c00 0x080>, + <0x30cc0e00 0x080>; + reg-names = "ram", "regs", "rxfifo", "txfifo"; + interrupts = <0x0 128 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_IPG>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_PHY>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_SPBA2_ROOT>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUDPLL_ROOT>; + clock-names = "ipg", "phy", "spba", "pll_ipg"; + dmas = <&sdma2 30 2 0>, <&sdma2 31 2 0>; + dma-names = "rx", "tx"; + resets = <&audiomix_reset 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index c60a5732d29c..8b4f4015cfe4 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -34,6 +34,18 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8960" + "fsl,imx-audio-mqs" + + "fsl,imx-audio-wm8524" + + "fsl,imx-audio-tlv320aic32x4" + + "fsl,imx-audio-tlv320aic31xx" + + "fsl,imx-audio-si476x" + + "fsl,imx-audio-wm8958" + Required properties: - compatible : Contains one of entries in the compatible list. @@ -44,6 +56,11 @@ Required properties: - audio-codec : The phandle of an audio codec +Optional properties: + + - audio-asrc : The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + - audio-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 @@ -60,10 +77,14 @@ Required properties: coexisting in order to support the old bindings of wm8962 and sgtl5000. -Optional properties: - - - audio-asrc : The phandle of ASRC. It can be absent if there's no - need to add ASRC support via DPCM. + - hp-det-gpio : The GPIO that detect headphones are plugged in + - mic-det-gpio : The GPIO that detect microphones are plugged in + - bitclock-master : Indicates dai-link bit clock master; for details see simple-card.yaml. + - frame-master : Indicates dai-link frame master; for details see simple-card.yaml. + - dai-format : audio format, for details see simple-card.yaml. + - frame-inversion : dai-link uses frame clock inversion, for details see simple-card.yaml. + - bitclock-inversion : dai-link uses bit clock inversion, for details see simple-card.yaml. + - mclk-id : main clock id, specific for each card configuration. Optional unless SSI is selected as a CPU DAI: diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 2e726b983845..c71c5861d787 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -8,7 +8,11 @@ codec/DSP interfaces. Required properties: - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai" or "fsl,imx6ul-sai" + "fsl,imx6sx-sai", "fsl,imx6ul-sai", + "fsl,imx7ulp-sai", "fsl,imx8mq-sai", + "fsl,imx8qm-sai", "fsl,imx8mm-sai", + "fsl,imx8mn-sai", "fsl,imx8mp-sai", or + "fsl,imx8ulp-sai". - reg : Offset and length of the register set for the device. diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt deleted file mode 100644 index 1084f7f22eea..000000000000 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt +++ /dev/null @@ -1,26 +0,0 @@ -* 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/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml new file mode 100644 index 000000000000..c3e9f3485449 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,cros-ec-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio codec controlled by ChromeOS EC + +maintainers: + - Cheng-Yi Chiang <cychiang@chromium.org> + - Tzung-Bi Shih <tzungbi@google.com> + +description: | + 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 inside the "codecs" + subnode of a cros-ec node. + (see Documentation/devicetree/bindings/mfd/google,cros-ec.yaml). + +properties: + compatible: + const: google,cros-ec-codec + + "#sound-dai-cells": + const: 1 + + reg: + items: + - description: | + Physical base address and length of shared memory region from EC. + It contains 3 unsigned 32-bit integer. The first 2 integers + combine to become an unsigned 64-bit physical address. + The last one integer is the length of the shared memory. + + memory-region: + maxItems: 1 + description: | + Shared memory region to EC. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + +required: + - compatible + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + reserved_mem: reserved-mem@52800000 { + compatible = "shared-dma-pool"; + reg = <0x52800000 0x100000>; + no-map; + }; + spi { + #address-cells = <1>; + #size-cells = <0>; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + reg = <0>; + + codecs { + #address-cells = <2>; + #size-cells = <1>; + + cros_ec_codec: ec-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml new file mode 100644 index 000000000000..233caa0ade87 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -0,0 +1,141 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7180-trogdor.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7180-Trogdor ASoC sound card driver + +maintainers: + - Rohit kumar <rohitkr@codeaurora.org> + - Cheng-Yi Chiang <cychiang@chromium.org> + +description: + This binding describes the SC7180 sound card which uses LPASS for audio. + +properties: + compatible: + enum: + - google,sc7180-trogdor + - google,sc7180-coachz + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + dmic-gpios: + maxItems: 1 + description: GPIO for switching between DMICs + +patternProperties: + "^dai-link(@[0-9])?$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: + maxItems: 1 + + required: + - link-name + - cpu + - codec + + additionalProperties: false + +required: + - compatible + - model + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + + - | + sound { + compatible = "google,sc7180-trogdor"; + model = "sc7180-rt5682-max98357a-2mic"; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR"; + + #address-cells = <1>; + #size-cells = <0>; + + dmic-gpios = <&tlmm 86 0>; + + dai-link@0 { + link-name = "MultiMedia0"; + reg = <0>; + cpu { + sound-dai = <&lpass_cpu 0>; + }; + + codec { + sound-dai = <&alc5682 0>; + }; + }; + + dai-link@1 { + link-name = "MultiMedia1"; + reg = <1>; + cpu { + sound-dai = <&lpass_cpu 1>; + }; + + codec { + sound-dai = <&max98357a>; + }; + }; + + dai-link@2 { + link-name = "MultiMedia2"; + reg = <2>; + cpu { + sound-dai = <&lpass_hdmi 0>; + }; + + codec { + sound-dai = <&msm_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml new file mode 100644 index 000000000000..869b40363af8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -0,0 +1,180 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7280-herobrine.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7280-Herobrine ASoC sound card driver + +maintainers: + - Srinivasa Rao Mandadapu <srivasam@codeaurora.org> + - Judy Hsiao <judyhsiao@chromium.org> + +description: + This binding describes the SC7280 sound card which uses LPASS for audio. + +properties: + compatible: + enum: + - google,sc7280-herobrine + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: true + + required: + - sound-dai + + additionalProperties: false + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: true + + required: + - sound-dai + + additionalProperties: false + + required: + - link-name + - cpu + - codec + - reg + + additionalProperties: false + +required: + - compatible + - model + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound { + compatible = "google,sc7280-herobrine"; + model = "sc7280-wcd938x-max98360a-4dmic"; + + audio-routing = + "IN1_HPHL", "HPHL_OUT", + "IN2_HPHR", "HPHR_OUT", + "AMIC1", "MIC BIAS1", + "AMIC2", "MIC BIAS2", + "VA DMIC0", "MIC BIAS3", + "VA DMIC1", "MIC BIAS3", + "VA DMIC2", "MIC BIAS4", + "VA DMIC3", "MIC BIAS4", + "TX SWR_ADC0", "ADC1_OUTPUT", + "TX SWR_ADC1", "ADC2_OUTPUT", + "TX SWR_ADC2", "ADC3_OUTPUT", + "TX SWR_DMIC0", "DMIC1_OUTPUT", + "TX SWR_DMIC1", "DMIC2_OUTPUT", + "TX SWR_DMIC2", "DMIC3_OUTPUT", + "TX SWR_DMIC3", "DMIC4_OUTPUT"; + + #address-cells = <1>; + #size-cells = <0>; + + dai-link@0 { + link-name = "WCD Playback"; + reg = <LPASS_CDC_DMA_RX0>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_RX0>; + }; + + codec { + sound-dai = <&wcd938x 0>, <&swr0 0>, <&rxmacro 0>; + }; + }; + dai-link@1 { + link-name = "WCD Capture"; + reg = <LPASS_CDC_DMA_TX3>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_TX3>; + }; + + codec { + sound-dai = <&wcd938x 1>, <&swr1 0>, <&txmacro 0>; + }; + }; + + dai-link@2 { + link-name = "MI2S Playback"; + reg = <MI2S_SECONDARY>; + cpu { + sound-dai = <&lpass_cpu MI2S_SECONDARY>; + }; + + codec { + sound-dai = <&max98360a>; + }; + }; + + dai-link@3 { + link-name = "DMIC Capture"; + reg = <LPASS_CDC_DMA_VA_TX0>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_VA_TX0>; + }; + + codec { + sound-dai = <&vamacro 0>; + }; + }; + + dai-link@5 { + link-name = "DP Playback"; + reg = <LPASS_DP_RX>; + cpu { + sound-dai = <&lpass_cpu LPASS_DP_RX>; + }; + + codec { + sound-dai = <&mdss_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt index 5efc8c068de0..efa32a486c4a 100644 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ b/Documentation/devicetree/bindings/sound/gtm601.txt @@ -1,10 +1,16 @@ GTM601 UMTS modem audio interface CODEC -This device has no configuration interface. Sample rate is fixed - 8kHz. +This device has no configuration interface. The sample rate and channels are +based on the compatible string + "option,gtm601" = 8kHz mono + "broadmobi,bm818" = 48KHz stereo Required properties: - - compatible : "option,gtm601" + - compatible : one of + "option,gtm601" + "broadmobi,bm818" + Example: diff --git a/Documentation/devicetree/bindings/sound/hdmi.txt b/Documentation/devicetree/bindings/sound/hdmi.txt deleted file mode 100644 index 56407c30e954..000000000000 --- a/Documentation/devicetree/bindings/sound/hdmi.txt +++ /dev/null @@ -1,16 +0,0 @@ -Device-Tree bindings for dummy HDMI codec - -Required properties: - - compatible: should be "linux,hdmi-audio". - -CODEC output pins: - * TX - -CODEC input pins: - * RX - -Example node: - - hdmi_audio: hdmi_audio@0 { - compatible = "linux,hdmi-audio"; - }; diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt index b02e3a6c0fef..e6f05f2f6c4e 100644 --- a/Documentation/devicetree/bindings/sound/ics43432.txt +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -1,4 +1,4 @@ -Invensense ICS-43432 MEMS microphone with I2S output. +Invensense ICS-43432-compatible MEMS microphone with I2S output. There are no software configuration options for this device, indeed, the only host connection is the I2S interface. Apart from requirements on clock @@ -8,7 +8,9 @@ contain audio data. A hardware pin determines if the device outputs data on the left or right channel of the I2S frame. Required properties: - - compatible : Must be "invensense,ics43432" + - compatible: should be one of the following. + "invensense,ics43432": For the Invensense ICS43432 + "cui,cmm-4030d-261": For the CUI CMM-4030D-261-I2S-TR Example: diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml new file mode 100644 index 000000000000..bb3a435722c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -0,0 +1,125 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio sound card. + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + format: + description: audio format. + items: + enum: + - i2s + - dsp_b + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: + minItems: 1 + maxItems: 2 + + fsl,mclk-equal-bclk: + description: Indicates mclk can be equal to bclk, especially for sai interface + $ref: /schemas/types.yaml#/definitions/flag + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - model + +additionalProperties: false + +examples: + - | + sound-ak4458 { + compatible = "fsl,imx-audio-card"; + model = "ak4458-audio"; + pri-dai-link { + link-name = "akcodec"; + format = "i2s"; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + fe-dai-link { + link-name = "HiFi-ASRC-FE"; + format = "i2s"; + cpu { + sound-dai = <&easrc>; + }; + }; + be-dai-link { + link-name = "HiFi-ASRC-BE"; + format = "dsp_b"; + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000000..e7e7bb65c366 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-hdmi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio complex with HDMI + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-hdmi + - fsl,imx-audio-sii902x + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + + hdmi-out: + type: boolean + description: | + This is a boolean property. If present, the transmitting function + of HDMI will be enabled, indicating there's a physical HDMI out + connector or jack on the board or it's connecting to some other IP + block, such as an HDMI encoder or display-controller. + + hdmi-in: + type: boolean + description: | + This is a boolean property. If present, the receiving function of + HDMI will be enabled, indicating there is a physical HDMI in + connector/jack on the board. + +required: + - compatible + - model + - audio-cpu + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "fsl,imx-audio-hdmi"; + model = "audio-hdmi"; + audio-cpu = <&aud2htx>; + hdmi-out; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt deleted file mode 100644 index 2db4dcbee1b9..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ /dev/null @@ -1,28 +0,0 @@ -Freescale Digital Audio Mux (AUDMUX) device - -Required properties: - - - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used - on i.MX21, or "fsl,imx31-audmux" for the version - firstly used on i.MX31. - - - reg : Should contain AUDMUX registers location and length. - -An initial configuration can be setup using child nodes. - -Required properties of optional child nodes: - - - fsl,audmux-port : Integer of the audmux port that is configured by this - child node. - - - fsl,port-config : List of configuration options for the specific port. - For imx31-audmux and above, it is a list of tuples - <ptcr pdcr>. For imx21-audmux it is a list of pcr - values. - -Example: - -audmux@21d8000 { - compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; - reg = <0x021d8000 0x4000>; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000000..dab45c310670 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel@pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,aic.yaml b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml new file mode 100644 index 000000000000..d607325f2f15 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml @@ -0,0 +1,92 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ingenic,aic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Ingenic SoCs AC97 / I2S Controller (AIC) DT bindings + +maintainers: + - Paul Cercueil <paul@crapouillou.net> + +properties: + $nodename: + pattern: '^audio-controller@' + + compatible: + oneOf: + - enum: + - ingenic,jz4740-i2s + - ingenic,jz4760-i2s + - ingenic,jz4770-i2s + - ingenic,jz4780-i2s + - items: + - const: ingenic,jz4725b-i2s + - const: ingenic,jz4740-i2s + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: AIC clock + - description: I2S clock + - description: EXT clock + - description: PLL/2 clock + + clock-names: + items: + - const: aic + - const: i2s + - const: ext + - const: pll half + + dmas: + items: + - description: DMA controller phandle and request line for I2S RX + - description: DMA controller phandle and request line for I2S TX + + dma-names: + items: + - const: rx + - const: tx + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> + aic: audio-controller@10020000 { + compatible = "ingenic,jz4740-i2s"; + reg = <0x10020000 0x38>; + + #sound-dai-cells = <0>; + + interrupt-parent = <&intc>; + interrupts = <18>; + + clocks = <&cgu JZ4740_CLK_AIC>, + <&cgu JZ4740_CLK_I2S>, + <&cgu JZ4740_CLK_EXT>, + <&cgu JZ4740_CLK_PLL_HALF>; + clock-names = "aic", "i2s", "ext", "pll half"; + + dmas = <&dmac 25 0xffffffff>, <&dmac 24 0xffffffff>; + dma-names = "rx", "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml new file mode 100644 index 000000000000..48aae54dd643 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ingenic,codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Ingenic JZ47xx internal codec DT bindings + +maintainers: + - Paul Cercueil <paul@crapouillou.net> + +properties: + $nodename: + pattern: '^audio-codec@.*' + + compatible: + oneOf: + - enum: + - ingenic,jz4770-codec + - ingenic,jz4760-codec + - ingenic,jz4725b-codec + - ingenic,jz4740-codec + - items: + - const: ingenic,jz4760b-codec + - const: ingenic,jz4760-codec + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: aic + + '#sound-dai-cells': + const: 0 + +additionalProperties: false + +required: + - compatible + - reg + - clocks + - clock-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> + 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/ingenic,jz4725b-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt deleted file mode 100644 index 05adc0d47b13..000000000000 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt +++ /dev/null @@ -1,20 +0,0 @@ -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 deleted file mode 100644 index 1ffcade87e7b..000000000000 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt +++ /dev/null @@ -1,20 +0,0 @@ -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/ingenic,jz4740-i2s.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt deleted file mode 100644 index b623d50004fb..000000000000 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt +++ /dev/null @@ -1,23 +0,0 @@ -Ingenic JZ4740 I2S controller - -Required properties: -- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s" -- reg : I2S registers location and length -- clocks : AIC and I2S PLL clock specifiers. -- clock-names: "aic" and "i2s" -- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels -- dma-names: Must be "tx" and "rx" - -Example: - -i2s: i2s@10020000 { - compatible = "ingenic,jz4740-i2s"; - reg = <0x10020000 0x94>; - - clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>; - clock-names = "aic", "i2s"; - - dmas = <&dma 2>, <&dma 3>; - dma-names = "tx", "rx"; - -}; diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml new file mode 100644 index 000000000000..803627e984f6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright 2020 Intel Corporation +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Intel KeemBay I2S Device Tree Bindings + +maintainers: + - Sia, Jee Heng <jee.heng.sia@intel.com> + +description: | + Intel KeemBay I2S + +properties: + compatible: + enum: + - intel,keembay-i2s + - intel,keembay-tdm + - intel,keembay-hdmi-i2s + + "#sound-dai-cells": + const: 0 + + reg: + items: + - description: I2S registers + - description: I2S gen configuration + + reg-names: + items: + - const: i2s-regs + - const: i2s_gen_cfg + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: osc + - const: apb_clk + + dmas: + items: + - description: DMA TX channel + - description: DMA RX channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #define KEEM_BAY_PSS_AUX_I2S3 + #define KEEM_BAY_PSS_I2S3 + i2s3: i2s@20140000 { + compatible = "intel,keembay-i2s"; + #sound-dai-cells = <0>; + reg = <0x20140000 0x200>, /* I2S registers */ + <0x202a00a4 0x4>; /* I2S gen configuration */ + reg-names = "i2s-regs", "i2s_gen_cfg"; + interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; + clock-names = "osc", "apb_clk"; + clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>; + dmas = <&axi_dma0 29>, <&axi_dma0 33>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml new file mode 100644 index 000000000000..e3a1f485f664 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bluetooth SCO Audio Codec Device Tree Bindings + +maintainers: + - Mark Brown <broonie@kernel.org> + +properties: + '#sound-dai-cells': + enum: + - 0 + + # For Wideband PCM + - 1 + + compatible: + enum: + - delta,dfbmcs320 + - linux,bt-sco + +required: + - '#sound-dai-cells' + - compatible + +additionalProperties: false + +examples: + - | + codec { + #sound-dai-cells = <0>; + compatible = "linux,bt-sco"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml new file mode 100644 index 000000000000..a4f9257e313d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,spdif-dit.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dummy SPDIF Transmitter Device Tree Bindings + +maintainers: + - Mark Brown <broonie@kernel.org> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: linux,spdif-dit + + "#sound-dai-cells": + const: 0 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + +additionalProperties: false + +examples: + - | + spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/madera.txt b/Documentation/devicetree/bindings/sound/madera.txt deleted file mode 100644 index 5e669ce552f4..000000000000 --- a/Documentation/devicetree/bindings/sound/madera.txt +++ /dev/null @@ -1,67 +0,0 @@ -Cirrus Logic Madera class audio codecs - -This describes audio configuration bindings for these codecs. - -See also the core bindings for the parent MFD driver: -See Documentation/devicetree/bindings/mfd/madera.txt - -and defines for values used in these bindings: -include/dt-bindings/sound/madera.h - -These properties are all contained in the parent MFD node. - -Optional properties: - - cirrus,dmic-ref : Indicates how the MICBIAS pins have been externally - connected to DMICs on each input, one cell per input. - <IN1 IN2 IN3 ...> - A value of 0 indicates MICVDD and is the default, other values depend on the - codec: - For CS47L35 one of the CS47L35_DMIC_REF_xxx values - For all other codecs one of the MADERA_DMIC_REF_xxx values - Also see the datasheet for a description of the INn_DMIC_SUP field. - - - cirrus,inmode : A list of input mode settings for each input. A maximum of - 16 cells, with four cells per input in the order INnAL, INnAR INnBL INnBR. - For non-muxed inputs the first two cells for that input set the mode for - the left and right channel and the second two cells must be 0. - For muxed inputs the first two cells for that input set the mode of the - left and right A inputs and the second two cells set the mode of the left - and right B inputs. - Valid mode values are one of the MADERA_INMODE_xxx. If the array is shorter - than the number of inputs the unspecified inputs default to - MADERA_INMODE_DIFF. - - - cirrus,out-mono : Mono bit for each output, maximum of six cells if the - array is shorter outputs will be set to stereo. - - - cirrus,max-channels-clocked : Maximum number of channels that I2S clocks - will be generated for. Useful when clock master for systems where the I2S - bus has multiple data lines. - One cell for each AIF, use a value of zero for AIFs that should be handled - normally. - - - cirrus,pdm-fmt : PDM speaker data format, must contain 2 cells - (OUT5 and OUT6). See the PDM_SPKn_FMT field in the datasheet for a - description of this value. - The second cell is ignored for codecs that do not have OUT6. - - - cirrus,pdm-mute : PDM mute format, must contain 2 cells - (OUT5 and OUT6). See the PDM_SPKn_CTRL_1 register in the datasheet for a - description of this value. - The second cell is ignored for codecs that do not have OUT6. - -Example: - -cs47l35@0 { - compatible = "cirrus,cs47l35"; - - cirrus,dmic-ref = <0 0 CS47L35_DMIC_REF_MICBIAS1B 0>; - cirrus,inmode = < - MADERA_INMODE_DMIC MADERA_INMODE_DMIC /* IN1A digital */ - MADERA_INMODE_SE MADERA_INMODE_SE /* IN1B single-ended */ - MADERA_INMODE_DIFF MADERA_INMODE_DIFF /* IN2 differential */ - 0 0 /* not used on this codec */ - >; - cirrus,out-mono = <0 0 0 0 0 0>; - cirrus,max-channels-clocked = <2 0 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml new file mode 100644 index 000000000000..81f266d66ec5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/marvell,mmp-sspa.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Marvel SSPA Digital Audio Interface Bindings + +maintainers: + - Lubomir Rintel <lkundrak@v3.sk> + +properties: + $nodename: + pattern: "^audio-controller(@.*)?$" + + compatible: + const: marvell,mmp-sspa + + reg: + items: + - description: RX block + - description: TX block + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Clock for the Audio block + - description: I2S bit clock + + clock-names: + items: + - const: audio + - const: bitclk + + power-domains: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + properties: + endpoint: + type: object + + properties: + dai-format: + const: i2s + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - port + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/marvell,mmp2.h> + + audio-controller@d42a0c00 { + compatible = "marvell,mmp-sspa"; + reg = <0xd42a0c00 0x30>, + <0xd42a0c80 0x30>; + interrupts = <2>; + clock-names = "audio", "bitclk"; + clocks = <&soc_clocks 127>, + <&audio_clk 1>; + #sound-dai-cells = <0>; + dmas = <&adma0 0>, <&adma0 1>; + dma-names = "tx", "rx"; + port { + endpoint { + remote-endpoint = <&rt5631_0>; + dai-format = "i2s"; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 7e1bbd5c27fd..39d640294c62 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -55,5 +55,5 @@ audio-codec@10 { compatible = "maxim,max98090"; reg = <0x10>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index 4bce14ce806f..75db84d06240 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -1,9 +1,10 @@ -Maxim MAX98357A audio DAC +Maxim MAX98357A/MAX98360A audio DAC -This node models the Maxim MAX98357A DAC. +This node models the Maxim MAX98357A/MAX98360A DAC. Required properties: -- compatible : "maxim,max98357a" +- compatible : "maxim,max98357a" for MAX98357A. + "maxim,max98360a" for MAX98360A. Optional properties: - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. @@ -20,3 +21,8 @@ max98357a { compatible = "maxim,max98357a"; sdmode-gpios = <&qcom_pinmux 25 0>; }; + +max98360a { + compatible = "maxim,max98360a"; + sdmode-gpios = <&qcom_pinmux 25 0>; +}; diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt index f6171591ddc6..98cb9ba5b328 100644 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -30,6 +30,9 @@ Required properties: - reg : the I2C address of the device for I2C +Optional properties: + - reset-gpios : GPIO to reset the device + Example: codec: max98927@3a { diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000000..deaa6886c42f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management + +maintainers: + - Steve Lee <steves.lee@maximintegrated.com> + +properties: + compatible: + const: maxim,max98390 + + reg: + maxItems: 1 + description: I2C address of the device. + + maxim,temperature_calib: + description: The calculated temperature data was measured while doing the calibration. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 65535 + + maxim,r0_calib: + description: This is r0 calibration data which was measured in factory mode. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 8388607 + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + reset-gpios = <&gpio 9 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..3f88c7d61e34 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song@maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml b/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml new file mode 100644 index 000000000000..0481315cb5f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,i2s-mcc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip I2S Multi-Channel Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and + supports a Time Division Multiplexed (TDM) interface with external + multi-channel audio codecs. It consists of a receiver, a transmitter and a + common clock generator that can be enabled separately to provide Adapter, + Client or Controller modes with receiver and/or transmitter active. + On later I2SMCC versions (starting with Microchip's SAMA7G5) I2S + multi-channel is supported by using multiple data pins, output and + input, without TDM. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - microchip,sam9x60-i2smcc + - microchip,sama7g5-i2smcc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock (Optional). Should be set mostly when Master + Mode is required. + minItems: 1 + + clock-names: + items: + - const: pclk + - const: gclk + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + microchip,tdm-data-pair: + description: + Represents the DIN/DOUT pair pins that are used to receive/send + TDM data. It is optional and it is only needed if the controller + uses the TDM mode. + $ref: /schemas/types.yaml#/definitions/uint8 + enum: [0, 1, 2, 3] + default: 0 + +if: + properties: + compatible: + const: microchip,sam9x60-i2smcc +then: + properties: + microchip,tdm-data-pair: false + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f001c000 { + #sound-dai-cells = <0>; + compatible = "microchip,sam9x60-i2smcc"; + reg = <0xf001c000 0x100>; + interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(36))>, + <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(37))>; + dma-names = "tx", "rx"; + clocks = <&i2s_clk>, <&i2s_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml new file mode 100644 index 000000000000..970311143253 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Rx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Receiver is a serial port + compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdifrx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdifrx: spdifrx@e1614000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdifrx"; + reg = <0xe1614000 0x4000>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(49)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 84>, <&pmc PMC_TYPE_GCK 84>; + clock-names = "pclk", "gclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000000..d5c022e49526 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a serial port + compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + maxItems: 1 + + dma-names: + const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt b/Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt deleted file mode 100644 index 91ec83a6faed..000000000000 --- a/Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt +++ /dev/null @@ -1,43 +0,0 @@ -* Microchip I2S Multi-Channel Controller - -Required properties: -- compatible: Should be "microchip,sam9x60-i2smcc". -- reg: Should be the physical base address of the controller and the - length of memory mapped region. -- interrupts: Should contain the interrupt for the controller. -- dmas: Should be one per channel name listed in the dma-names property, - as described in atmel-dma.txt and dma.txt files. -- dma-names: Identifier string for each DMA request line in the dmas property. - Two dmas have to be defined, "tx" and "rx". -- clocks: Must contain an entry for each entry in clock-names. - Please refer to clock-bindings.txt. -- clock-names: Should be one of each entry matching the clocks phandles list: - - "pclk" (peripheral clock) Required. - - "gclk" (generated clock) Optional (1). - -Optional properties: -- pinctrl-0: Should specify pin control groups used for this controller. -- princtrl-names: Should contain only one value - "default". - - -(1) : Only the peripheral clock is required. The generated clock is optional - and should be set mostly when Master Mode is required. - -Example: - - i2s@f001c000 { - compatible = "microchip,sam9x60-i2smcc"; - reg = <0xf001c000 0x100>; - interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>; - dmas = <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(36))>, - <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(37))>; - dma-names = "tx", "rx"; - clocks = <&i2s_clk>, <&i2s_gclk>; - clock-names = "pclk", "gclk"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s_default>; - }; diff --git a/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml new file mode 100644 index 000000000000..04414eb4ada9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,pdmc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip Pulse Density Microphone Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Pulse Density Microphone Controller (PDMC) interfaces up to 4 + digital microphones having Pulse Density Modulated (PDM) outputs. + +properties: + compatible: + const: microchip,sama7g5-pdmc + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + + microchip,mic-pos: + description: | + Position of PDM microphones on the DS line and the sampling edge (rising + or falling) of the CLK line. A microphone is represented as a pair of DS + line and the sampling edge. The first microphone is mapped to channel 0, + the second to channel 1, etc. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: value for DS line + - description: value for sampling edge + anyOf: + - enum: + - [0, 0] + - [0, 1] + - [1, 0] + - [1, 1] + minItems: 1 + maxItems: 4 + uniqueItems: true + +required: + - compatible + - reg + - "#sound-dai-cells" + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - microchip,mic-pos + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/sound/microchip,pdmc.h> + + pdmc: sound@e1608000 { + compatible = "microchip,sama7g5-pdmc"; + reg = <0xe1608000 0x4000>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 68 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(37)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 68>, <&pmc PMC_TYPE_GCK 68>; + clock-names = "pclk", "gclk"; + microchip,mic-pos = <MCHP_PDMC_DS0 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS0 MCHP_PDMC_CLK_NEGATIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_NEGATIVE>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt6358.txt b/Documentation/devicetree/bindings/sound/mt6358.txt index 5465730013a1..59a73ffdf1d3 100644 --- a/Documentation/devicetree/bindings/sound/mt6358.txt +++ b/Documentation/devicetree/bindings/sound/mt6358.txt @@ -10,9 +10,15 @@ Required properties: - compatible : "mediatek,mt6358-sound". - Avdd-supply : power source of AVDD +Optional properties: +- mediatek,dmic-mode : Indicates how many data pins are used to transmit two + channels of PDM signal. 0 means two wires, 1 means one wire. Default + value is 0. + Example: mt6358_snd { compatible = "mediatek,mt6358-sound"; Avdd-supply = <&mt6358_vaud28_reg>; + mediatek,dmic-mode = <0>; }; diff --git a/Documentation/devicetree/bindings/sound/mt6359.yaml b/Documentation/devicetree/bindings/sound/mt6359.yaml new file mode 100644 index 000000000000..a54f466f769d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6359.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt6359.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek MT6359 Codec Device Tree Bindings + +maintainers: + - Eason Yen <eason.yen@mediatek.com> + - Jiaxin Yu <jiaxin.yu@mediatek.com> + - Shane Chien <shane.chien@mediatek.com> + +description: | + The communication between MT6359 and SoC is through Mediatek PMIC wrapper. + For more detail, please visit Mediatek PMIC wrapper documentation. + Must be a child node of PMIC wrapper. + +properties: + mediatek,dmic-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Indicates how many data pins are used to transmit two channels of PDM + signal. 0 means two wires, 1 means one wire. Default value is 0. + enum: + - 0 # one wire + - 1 # two wires + + mediatek,mic-type-0: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Specifies the type of mic type connected to adc0 + + enum: + - 0 # IDLE - mic in turn-off status + - 1 # ACC - analog mic with alternating coupling + - 2 # DMIC - digital mic + - 3 # DCC - analog mic with direct couping + - 4 # DCC_ECM_DIFF - analog electret condenser mic with differential mode + - 5 # DCC_ECM_SINGLE - analog electret condenser mic with single mode + + mediatek,mic-type-1: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Specifies the type of mic type connected to adc1 + + mediatek,mic-type-2: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Specifies the type of mic type connected to adc2 + +additionalProperties: false + +examples: + - | + mt6359codec: mt6359codec { + mediatek,dmic-mode = <0>; + mediatek,mic-type-0 = <2>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt index 396ba38619f6..1f1cba4152ce 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt @@ -4,6 +4,10 @@ Required properties: - compatible = "mediatek,mt68183-audio"; - reg: register location and size - interrupts: should contain AFE interrupt +- resets: Must contain an entry for each entry in reset-names + See ../reset/reset.txt for details. +- reset-names: should have these reset names: + "audiosys"; - 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: @@ -20,6 +24,8 @@ Example: compatible = "mediatek,mt8183-audio"; reg = <0 0x11220000 0 0x1000>; interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>; + resets = <&watchdog MT8183_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; power-domains = <&scpsys MT8183_POWER_DOMAIN_AUDIO>; clocks = <&infrasys CLK_INFRA_AUDIO>, <&infrasys CLK_INFRA_AUDIO_26M_BCLK>, diff --git a/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt index 92ac86f83822..f276dfc74b46 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt @@ -1,15 +1,21 @@ -MT8183 with MT6358, DA7219 and MAX98357 CODECS +MT8183 with MT6358, DA7219, MAX98357, and RT1015 CODECS Required properties: -- compatible : "mediatek,mt8183_da7219_max98357" +- compatible : "mediatek,mt8183_da7219_max98357" for MAX98357A codec + "mediatek,mt8183_da7219_rt1015" for RT1015 codec + "mediatek,mt8183_da7219_rt1015p" for RT1015P codec - mediatek,headset-codec: the phandles of da7219 codecs - mediatek,platform: the phandle of MT8183 ASoC platform +Optional properties: +- mediatek,hdmi-codec: the phandles of HDMI codec + Example: sound { compatible = "mediatek,mt8183_da7219_max98357"; mediatek,headset-codec = <&da7219>; + mediatek,hdmi-codec = <&it6505dptx>; mediatek,platform = <&afe>; }; diff --git a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt index d6d5207fa996..ecd46ed8eb98 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -1,15 +1,25 @@ -MT8183 with MT6358, TS3A227 and MAX98357 CODECS +MT8183 with MT6358, TS3A227, MAX98357, and RT1015 CODECS Required properties: -- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" -- mediatek,headset-codec: the phandles of ts3a227 codecs +- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" for MAX98357A codec + "mediatek,mt8183_mt6358_ts3a227_max98357b" for MAX98357B codec + "mediatek,mt8183_mt6358_ts3a227_rt1015" for RT1015 codec + "mediatek,mt8183_mt6358_ts3a227_rt1015p" for RT1015P codec - mediatek,platform: the phandle of MT8183 ASoC platform +Optional properties: +- mediatek,headset-codec: the phandles of ts3a227 codecs +- mediatek,ec-codec: the phandle of EC codecs. + See google,cros-ec-codec.txt for more details. +- mediatek,hdmi-codec: the phandles of HDMI codec + Example: sound { compatible = "mediatek,mt8183_mt6358_ts3a227_max98357"; mediatek,headset-codec = <&ts3a227>; + mediatek,ec-codec = <&ec_codec>; + mediatek,hdmi-codec = <&it6505dptx>; mediatek,platform = <&afe>; }; diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..7a25bc9b8060 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu@mediatek.com> + - Shane Chien <shane.chien@mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml new file mode 100644 index 000000000000..4fa179909c62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-mt6359-rt1015-rt5682.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek MT8192 with MT6359, RT1015 and RT5682 ASoC sound card driver + +maintainers: + - Jiaxin Yu <jiaxin.yu@mediatek.com> + - Shane Chien <shane.chien@mediatek.com> + +description: + This binding describes the MT8192 sound card. + +properties: + compatible: + enum: + - mediatek,mt8192_mt6359_rt1015_rt5682 + - mediatek,mt8192_mt6359_rt1015p_rt5682 + - mediatek,mt8192_mt6359_rt1015p_rt5682s + + mediatek,platform: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8192 ASoC platform. + + mediatek,hdmi-codec: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of HDMI codec. + + headset-codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle + required: + - sound-dai + + speaker-codecs: + type: object + properties: + sound-dai: + minItems: 1 + maxItems: 2 + items: + maxItems: 1 + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - sound-dai + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - speaker-codecs + +examples: + - | + + sound: mt8192-sound { + compatible = "mediatek,mt8192_mt6359_rt1015_rt5682"; + mediatek,platform = <&afe>; + mediatek,hdmi-codec = <&anx_bridge_dp>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + + headset-codec { + sound-dai = <&rt5682>; + }; + + speaker-codecs { + sound-dai = <&rt1015_l>, + <&rt1015_r>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml new file mode 100644 index 000000000000..4452a4070eff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml @@ -0,0 +1,200 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8195-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8195 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8195-audio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + memory-region: + maxItems: 1 + description: | + Shared memory region for AFE memif. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: 26M clock + - description: audio pll1 clock + - description: audio pll2 clock + - description: clock divider for i2si1_mck + - description: clock divider for i2si2_mck + - description: clock divider for i2so1_mck + - description: clock divider for i2so2_mck + - description: clock divider for dptx_mck + - description: a1sys hoping clock + - description: audio intbus clock + - description: audio hires clock + - description: audio local bus clock + - description: mux for dptx_mck + - description: mux for i2so1_mck + - description: mux for i2so2_mck + - description: mux for i2si1_mck + - description: mux for i2si2_mck + - description: audio infra 26M clock + - description: infra bus clock + + clock-names: + items: + - const: clk26m + - const: apll1_ck + - const: apll2_ck + - const: apll12_div0 + - const: apll12_div1 + - const: apll12_div2 + - const: apll12_div3 + - const: apll12_div9 + - const: a1sys_hp_sel + - const: aud_intbus_sel + - const: audio_h_sel + - const: audio_local_bus_sel + - const: dptx_m_sel + - const: i2so1_m_sel + - const: i2so2_m_sel + - const: i2si1_m_sel + - const: i2si2_m_sel + - const: infra_ao_audio_26m_b + - const: scp_adsp_audiodsp + + mediatek,etdm-in1-chn-disabled: + $ref: /schemas/types.yaml#/definitions/uint8-array + maxItems: 24 + description: Specify which input channel should be disabled. + + mediatek,etdm-in2-chn-disabled: + $ref: /schemas/types.yaml#/definitions/uint8-array + maxItems: 16 + description: Specify which input channel should be disabled. + +patternProperties: + "^mediatek,etdm-in[1-2]-mclk-always-on-rate-hz$": + description: Specify etdm in mclk output rate for always on case. + + "^mediatek,etdm-out[1-3]-mclk-always-on-rate-hz$": + description: Specify etdm out mclk output rate for always on case. + + "^mediatek,etdm-in[1-2]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-out[1-3]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-in[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in moudule. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + "^mediatek,etdm-out[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out moudule. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + +required: + - compatible + - reg + - interrupts + - resets + - reset-names + - mediatek,topckgen + - power-domains + - clocks + - clock-names + - memory-region + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe: mt8195-afe-pcm@10890000 { + compatible = "mediatek,mt8195-audio"; + reg = <0x10890000 0x10000>; + interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; + resets = <&watchdog 14>; + reset-names = "audiosys"; + mediatek,topckgen = <&topckgen>; + power-domains = <&spm 7>; //MT8195_POWER_DOMAIN_AUDIO + memory-region = <&snd_dma_mem_reserved>; + clocks = <&clk26m>, + <&topckgen 163>, //CLK_TOP_APLL1 + <&topckgen 166>, //CLK_TOP_APLL2 + <&topckgen 233>, //CLK_TOP_APLL12_DIV0 + <&topckgen 234>, //CLK_TOP_APLL12_DIV1 + <&topckgen 235>, //CLK_TOP_APLL12_DIV2 + <&topckgen 236>, //CLK_TOP_APLL12_DIV3 + <&topckgen 238>, //CLK_TOP_APLL12_DIV9 + <&topckgen 100>, //CLK_TOP_A1SYS_HP_SEL + <&topckgen 33>, //CLK_TOP_AUD_INTBUS_SEL + <&topckgen 34>, //CLK_TOP_AUDIO_H_SEL + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS_SEL + <&topckgen 98>, //CLK_TOP_DPTX_M_SEL + <&topckgen 94>, //CLK_TOP_I2SO1_M_SEL + <&topckgen 95>, //CLK_TOP_I2SO2_M_SEL + <&topckgen 96>, //CLK_TOP_I2SI1_M_SEL + <&topckgen 97>, //CLK_TOP_I2SI2_M_SEL + <&infracfg_ao 50>, //CLK_INFRA_AO_AUDIO_26M_B + <&scp_adsp 0>; //CLK_SCP_ADSP_AUDIODSP + clock-names = "clk26m", + "apll1_ck", + "apll2_ck", + "apll12_div0", + "apll12_div1", + "apll12_div2", + "apll12_div3", + "apll12_div9", + "a1sys_hp_sel", + "aud_intbus_sel", + "audio_h_sel", + "audio_local_bus_sel", + "dptx_m_sel", + "i2so1_m_sel", + "i2so2_m_sel", + "i2si1_m_sel", + "i2si2_m_sel", + "infra_ao_audio_26m_b", + "scp_adsp_audiodsp"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml new file mode 100644 index 000000000000..ad3447ff8b2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8195-mt6359.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8195 ASoC sound card driver + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +description: + This binding describes the MT8195 sound card. + +properties: + compatible: + enum: + - mediatek,mt8195_mt6359_rt1019_rt5682 + - mediatek,mt8195_mt6359_rt1011_rt5682 + - mediatek,mt8195_mt6359_max98390_rt5682 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mediatek,platform: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 ASoC platform. + + mediatek,dptx-codec: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 Display Port Tx codec node. + + mediatek,hdmi-codec: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 HDMI codec node. + + mediatek,adsp: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 ADSP platform. + + mediatek,dai-link: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +additionalProperties: false + +required: + - compatible + - mediatek,platform + +examples: + - | + + sound: mt8195-sound { + compatible = "mediatek,mt8195_mt6359_rt1019_rt5682"; + mediatek,platform = <&afe>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/name-prefix.txt b/Documentation/devicetree/bindings/sound/name-prefix.txt deleted file mode 100644 index 645775908657..000000000000 --- a/Documentation/devicetree/bindings/sound/name-prefix.txt +++ /dev/null @@ -1,24 +0,0 @@ -Name prefix: - -Card implementing the routing property define the connection between -audio components as list of string pair. Component using the same -sink/source names may use the name prefix property to prepend the -name of their sinks/sources with the provided string. - -Optional name prefix property: -- sound-name-prefix : string using as prefix for the sink/source names of - the component. - -Example: Two instances of the same component. - -amp0: analog-amplifier@0 { - compatible = "simple-audio-amplifier"; - enable-gpios = <&gpio GPIOH_3 0>; - sound-name-prefix = "FRONT"; -}; - -amp1: analog-amplifier@1 { - compatible = "simple-audio-amplifier"; - enable-gpios = <&gpio GPIOH_4 0>; - sound-name-prefix = "BACK"; -}; diff --git a/Documentation/devicetree/bindings/sound/name-prefix.yaml b/Documentation/devicetree/bindings/sound/name-prefix.yaml new file mode 100644 index 000000000000..2fe57f87ac52 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/name-prefix.yaml @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/name-prefix.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Component sound name prefix + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: | + Card implementing the routing property define the connection between + audio components as list of string pair. Component using the same + sink/source names may use this property to prepend the name of their + sinks/sources with the provided string. + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt new file mode 100644 index 000000000000..6eaec46f384c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8315.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8315 Mono Class-D Amplifier + +Required properties: +- compatible : "nuvoton,nau8315" + +Optional properties: +- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +#include <dt-bindings/gpio/gpio.h> + +nau8315 { + compatible = "nuvoton,nau8315"; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt index 05830e477acd..7deaa452b200 100644 --- a/Documentation/devicetree/bindings/sound/nau8810.txt +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -1,10 +1,11 @@ -NAU8810 audio CODEC +NAU8810/NAU8812/NAU8814 audio CODEC This device supports I2C only. Required properties: - - compatible : "nuvoton,nau8810" + - compatible : One of "nuvoton,nau8810" or "nuvoton,nau8812" or + "nuvoton,nau8814" - reg : the I2C address of the device. diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt new file mode 100644 index 000000000000..6c3baf7a5f21 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -0,0 +1,55 @@ +Nuvoton NAU88L21 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8821" + + - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - 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,dmic-clk-threshold: the ADC threshold of DMIC clock. + + +Example: + + headset: nau8821@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,dmic-clk-threshold = 3072000; + }; diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index d16d96839bcb..388a7bc60b1f 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -101,5 +101,5 @@ Example: nuvoton,crosstalk-enable; clock-names = "mclk"; - clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; + clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>; }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt deleted file mode 100644 index 57f40f93453e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ /dev/null @@ -1,48 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-alc5632" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the ALC5632's pins as documented in the binding for the device - and: - - * Headset Stereophone - * Int Spk - * Headset Mic - * Digital Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller -- nvidia,audio-codec : The phandle of the ALC5632 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-alc5632-paz00", - "nvidia,tegra-audio-alc5632"; - - nvidia,model = "Compal PAZ00"; - - nvidia,audio-routing = - "Int Spk", "SPK_OUTP", - "Int Spk", "SPK_OUTN", - "Headset Mic","MICBIAS1", - "MIC1_N", "Headset Mic", - "MIC1_P", "Headset Mic", - "Headset Stereophone", "HP_OUT_R", - "Headset Stereophone", "HP_OUT_L"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&alc5632>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml new file mode 100644 index 000000000000..7ef774910e5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-alc5632.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with ALC5632 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-alc5632(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-alc5632 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headset Stereophone" + - "Int Spk" + - "Headset Mic" + - "Digital Mic" + + # CODEC Pins + - SPKOUT + - SPKOUTN + - MICBIAS1 + - MIC1 + - HPR + - HPL + - DMICDAT + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "Headset Mic", "MICBIAS1", + "MIC1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL", + "DMICDAT", "Digital Mic"; + + nvidia,i2s-controller = <&i2s>; + nvidia,audio-codec = <&codec>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml new file mode 100644 index 000000000000..82801b4f46dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/nvidia,tegra-audio-common.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Common properties for NVIDIA Tegra audio complexes + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +properties: + clocks: + items: + - description: PLL A clock + - description: PLL A OUT0 clock + - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk + + clock-names: + items: + - const: pll_a + - const: pll_a_out0 + - const: mclk + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + + nvidia,ac97-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the AC97 controller + + nvidia,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the Tegra I2S controller + + nvidia,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,ear-sel-gpios: + maxItems: 1 + description: The GPIO that switch between the microphones + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml new file mode 100644 index 000000000000..b4bee466d67a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -0,0 +1,199 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-graph-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph based Tegra sound card driver + +description: | + This is based on generic audio graph card driver along with additional + customizations for Tegra platforms. It uses the same bindings with + additional standard clock DT bindings required for Tegra. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: audio-graph.yaml# + +properties: + compatible: + enum: + - nvidia,tegra210-audio-graph-card + - nvidia,tegra186-audio-graph-card + + clocks: + minItems: 2 + + clock-names: + items: + - const: pll_a + - const: plla_out0 + + assigned-clocks: + minItems: 1 + maxItems: 3 + + assigned-clock-parents: + minItems: 1 + maxItems: 3 + + assigned-clock-rates: + minItems: 1 + maxItems: 3 + + interconnects: + items: + - description: APE read memory client + - description: APE write memory client + + interconnect-names: + items: + - const: dma-mem # read + - const: write + + iommus: + maxItems: 1 + +required: + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +unevaluatedProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + tegra_sound { + compatible = "nvidia,tegra210-audio-graph-card"; + + clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + clock-names = "pll_a", "plla_out0"; + + assigned-clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA210_CLK_EXTERN1>; + assigned-clock-parents = <0>, <0>, <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <368640000>, <49152000>, <12288000>; + + dais = /* FE */ + <&admaif1_port>, + /* Router */ + <&xbar_i2s1_port>, + /* I/O DAP Ports */ + <&i2s1_port>; + + label = "jetson-tx1-ape"; + }; + + // The ports are defined for AHUB and its child devices. + ahub@702d0800 { + compatible = "nvidia,tegra210-ahub"; + reg = <0x702d0800 0x800>; + clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + clock-names = "ahub"; + assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0x702d0000 0x702d0000 0x0000e400>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + xbar_admaif1_ep: endpoint { + remote-endpoint = <&admaif1_ep>; + }; + }; + + // ... + + xbar_i2s1_port: port@a { + reg = <0xa>; + xbar_i2s1_ep: endpoint { + remote-endpoint = <&i2s1_cif_ep>; + }; + }; + }; + + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + admaif1_port: port@0 { + reg = <0x0>; + admaif1_ep: endpoint { + remote-endpoint = <&xbar_admaif1_ep>; + }; + }; + + // More ADMAIF ports to follow + }; + }; + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + reg = <0x702d1000 0x100>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + + i2s1_cif_ep: endpoint { + remote-endpoint = <&xbar_i2s1_ep>; + }; + }; + + i2s1_port: port@1 { + reg = <0x1>; + + i2s1_dap: endpoint { + dai-format = "i2s"; + }; + }; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt deleted file mode 100644 index c3495beba358..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ /dev/null @@ -1,53 +0,0 @@ -NVIDIA Tegra audio complex, with MAX98090 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-max98090" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the MAX98090's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the MAX98090 audio codec. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-max98090-venice2", - "nvidia,tegra-audio-max98090"; - nvidia,model = "NVIDIA Tegra Venice2"; - - nvidia,audio-routing = - "Headphones", "HPR", - "Headphones", "HPL", - "Speakers", "SPKR", - "Speakers", "SPKL", - "Mic Jack", "MICBIAS", - "IN34", "Mic Jack"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&acodec>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml new file mode 100644 index 000000000000..ccc2ee77ca30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml @@ -0,0 +1,97 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with MAX98090 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98090(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-max98090 + - items: + - enum: + - nvidia,tegra-audio-max98090-nyan-big + - nvidia,tegra-audio-max98090-nyan-blaze + - const: nvidia,tegra-audio-max98090-nyan + - const: nvidia,tegra-audio-max98090 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphones" + - "Speakers" + - "Mic Jack" + - "Int Mic" + + # CODEC Pins + - MIC1 + - MIC2 + - DMICL + - DMICR + - IN1 + - IN2 + - IN3 + - IN4 + - IN5 + - IN6 + - IN12 + - IN34 + - IN56 + - HPL + - HPR + - SPKL + - SPKR + - RCVL + - RCVR + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra124-car.h> + + sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt deleted file mode 100644 index 7788808dcd0b..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ /dev/null @@ -1,52 +0,0 @@ -NVIDIA Tegra audio complex, with RT5640 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5640" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the RT5640's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5640-dalmore", - "nvidia,tegra-audio-rt5640"; - nvidia,model = "NVIDIA Tegra Dalmore"; - - nvidia,audio-routing = - "Headphones", "HPOR", - "Headphones", "HPOL", - "Speakers", "SPORP", - "Speakers", "SPORN", - "Speakers", "SPOLP", - "Speakers", "SPOLN"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5640>; - - nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */ - - clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml new file mode 100644 index 000000000000..b1deaf271afa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5640.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5639 or RT5640 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt56(39|40)(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5640 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphones" + - "Speakers" + - "Mic Jack" + + # CODEC Pins + - DMIC1 + - DMIC2 + - MICBIAS1 + - IN1P + - IN1R + - IN2P + - IN2R + - HPOL + - HPOR + - LOUTL + - LOUTR + - MONOP + - MONON + - SPOLP + - SPOLN + - SPORP + - SPORN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5640-dalmore", + "nvidia,tegra-audio-rt5640"; + nvidia,model = "NVIDIA Tegra Dalmore"; + + nvidia,audio-routing = + "Headphones", "HPOR", + "Headphones", "HPOL", + "Speakers", "SPORP", + "Speakers", "SPORN", + "Speakers", "SPOLP", + "Speakers", "SPOLN"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5640>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, <&clk 217>, <&clk 120>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt deleted file mode 100644 index a4589cda214e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt +++ /dev/null @@ -1,67 +0,0 @@ -NVIDIA Tegra audio complex, with RT5677 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5677" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the RT5677's pins (as documented in its binding), and the jacks - on the board: - - * Headphone - * Speaker - * Headset Mic - * Internal Mic 1 - * Internal Mic 2 - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in -- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier -- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in -- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5677-ryu", - "nvidia,tegra-audio-rt5677"; - nvidia,model = "NVIDIA Tegra Ryu"; - - nvidia,audio-routing = - "Headphone", "LOUT2", - "Headphone", "LOUT1", - "Headset Mic", "MICBIAS1", - "IN1P", "Headset Mic", - "IN1N", "Headset Mic", - "DMIC L1", "Internal Mic 1", - "DMIC R1", "Internal Mic 1", - "DMIC L2", "Internal Mic 2", - "DMIC R2", "Internal Mic 2", - "Speaker", "PDM1L", - "Speaker", "PDM1R"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5677>; - - nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; - nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; - nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; - nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml new file mode 100644 index 000000000000..a49997d6028b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5677.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5677 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5677(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5677 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone" + - "Speaker" + - "Headset Mic" + - "Internal Mic 1" + - "Internal Mic 2" + + # CODEC Pins + - IN1P + - IN1N + - IN2P + - IN2N + - MICBIAS1 + - DMIC1 + - DMIC2 + - DMIC3 + - DMIC4 + - "DMIC L1" + - "DMIC L2" + - "DMIC L3" + - "DMIC L4" + - "DMIC R1" + - "DMIC R2" + - "DMIC R3" + - "DMIC R4" + - LOUT1 + - LOUT2 + - LOUT3 + - PDM1L + - PDM1R + - PDM2L + - PDM2R + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, + <&clk 217>, + <&clk 121>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt deleted file mode 100644 index 5da7da4ea07a..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt +++ /dev/null @@ -1,42 +0,0 @@ -NVIDIA Tegra audio complex, with SGTL5000 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-sgtl5000" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the SGTL5000's pins (as documented in its binding), and the jacks - on the board: - - * Headphone Jack - * Line In Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the SGTL5000 audio codec. - -Example: - -sound { - compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", - "nvidia,tegra-audio-sgtl5000"; - nvidia,model = "Toradex Apalis T30"; - nvidia,audio-routing = - "Headphone Jack", "HP_OUT", - "LINE_IN", "Line In Jack", - "MIC_IN", "Mic Jack"; - nvidia,i2s-controller = <&tegra_i2s2>; - nvidia,audio-codec = <&sgtl5000>; - clocks = <&tegra_car TEGRA30_CLK_PLL_A>, - <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA30_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml new file mode 100644 index 000000000000..943e7c01741c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with SGTL5000 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-sgtl5000([-_][a-z0-9]+)+$' + - const: nvidia,tegra-audio-sgtl5000 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Line In Jack" + - "Mic Jack" + + # CODEC Pins + - HP_OUT + - LINE_OUT + - LINE_IN + - MIC_IN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30 SGTL5000"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt deleted file mode 100644 index ef1fe7358279..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt +++ /dev/null @@ -1,21 +0,0 @@ -NVIDIA Tegra audio complex for TrimSlice - -Required properties: -- compatible : "nvidia,tegra-audio-trimslice" -- clocks : Must contain an entry for each entry in clock-names. -- clock-names : Must include the following entries: - "pll_a" (The Tegra clock of that name), - "pll_a_out0" (The Tegra clock of that name), - "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-trimslice"; - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&codec>; - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml new file mode 100644 index 000000000000..8c87cd166238 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-trimslice.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with TrimSlice CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + const: nvidia,tegra-audio-trimslice + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-trimslice"; + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt deleted file mode 100644 index 96f6a57dd6b4..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt +++ /dev/null @@ -1,40 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8753" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the WM8753's pins as documented in the binding for the WM8753, - and the jacks on the board: - - * Headphone Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8753 audio codec -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8753-whistler", - "nvidia,tegra-audio-wm8753" - nvidia,model = "tegra-wm8753-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "LOUT1", - "Headphone Jack", "ROUT1"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8753>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml new file mode 100644 index 000000000000..a5b431d7d0c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8753.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8753 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8753(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8753 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Mic Jack" + + # CODEC Pins + - LOUT1 + - LOUT2 + - ROUT1 + - ROUT2 + - MONO1 + - MONO2 + - OUT3 + - OUT4 + - LINE1 + - LINE2 + - RXP + - RXN + - ACIN + - ACOP + - MIC1N + - MIC1 + - MIC2N + - MIC2 + - "Mic Bias" + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8753-whistler", + "nvidia,tegra-audio-wm8753"; + nvidia,model = "tegra-wm8753-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "LOUT1", + "Headphone Jack", "ROUT1"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8753>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt deleted file mode 100644 index b795d282818d..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ /dev/null @@ -1,60 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8903" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the WM8903's pins (documented in the WM8903 binding document), - and the jacks on the board: - - * Headphone Jack - * Int Spk - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Optional properties: -- nvidia,spkr-en-gpios : The GPIO that enables the speakers -- nvidia,hp-mute-gpios : The GPIO that mutes the headphones -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone -- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8903-harmony", - "nvidia,tegra-audio-wm8903" - nvidia,model = "tegra-wm8903-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "HPOUTR", - "Headphone Jack", "HPOUTL", - "Int Spk", "ROP", - "Int Spk", "RON", - "Int Spk", "LOP", - "Int Spk", "LON", - "Mic Jack", "MICBIAS", - "IN1L", "Mic Jack"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8903>; - - nvidia,spkr-en-gpios = <&codec 2 0>; - nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */ - nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */ - nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */ - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml new file mode 100644 index 000000000000..1b836acab980 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8903.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8903 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8903(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8903 + - items: + - pattern: ad,tegra-audio-plutux + - const: nvidia,tegra-audio-wm8903 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Int Spk" + - "Mic Jack" + - "Int Mic" + + # CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - DMICDAT + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - LOP + - LON + - ROP + - RON + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8903-harmony", + "nvidia,tegra-audio-wm8903"; + nvidia,model = "tegra-wm8903-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "HPOUTR", + "Headphone Jack", "HPOUTL", + "Int Spk", "ROP", + "Int Spk", "RON", + "Int Spk", "LOP", + "Int Spk", "LON", + "Mic Jack", "MICBIAS", + "IN1L", "Mic Jack"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8903>; + + nvidia,spkr-en-gpios = <&codec 2 0>; + nvidia,hp-det-gpios = <&gpio 178 0>; + nvidia,int-mic-en-gpios = <&gpio 184 0>; + nvidia,ext-mic-en-gpios = <&gpio 185 0>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt deleted file mode 100644 index 436f6cd9d07c..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt +++ /dev/null @@ -1,60 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm9712" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-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 and - sinks are the WM9712's pins, and the jacks on the board: - - WM9712 pins: - - * MONOOUT - * HPOUTL - * HPOUTR - * LOUT2 - * ROUT2 - * OUT3 - * LINEINL - * LINEINR - * PHONE - * PCBEEP - * MIC1 - * MIC2 - * Mic Bias - - Board connectors: - - * Headphone - * LineIn - * Mic - -- nvidia,ac97-controller : The phandle of the Tegra AC97 controller - - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm9712-colibri_t20", - "nvidia,tegra-audio-wm9712"; - nvidia,model = "Toradex Colibri T20"; - - nvidia,audio-routing = - "Headphone", "HPOUTL", - "Headphone", "HPOUTR", - "LineIn", "LINEINL", - "LineIn", "LINEINR", - "Mic", "MIC1"; - - nvidia,ac97-controller = <&ac97>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml new file mode 100644 index 000000000000..a1448283344b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm9712.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM9712 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm9712([-_][a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm9712 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone" + - "LineIn" + - "Mic" + + # CODEC Pins + - MONOOUT + - HPOUTL + - HPOUTR + - LOUT2 + - ROUT2 + - OUT3 + - LINEINL + - LINEINR + - PHONE + - PCBEEP + - MIC1 + - MIC2 + - "Mic Bias" + +required: + - nvidia,ac97-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm9712-colibri_t20", + "nvidia,tegra-audio-wm9712"; + nvidia,model = "Toradex Colibri T20"; + + nvidia,audio-routing = + "Headphone", "HPOUTL", + "Headphone", "HPOUTR", + "LineIn", "LINEINL", + "LineIn", "LINEINR", + "Mic", "MIC1"; + + nvidia,ac97-controller = <&ac97>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml new file mode 100644 index 000000000000..520d0d063d1a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra186-asrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra186 ASRC Device Tree Bindings + +description: | + Asynchronous Sample Rate Converter (ASRC) converts the sampling frequency + of the input signal from one frequency to another. It can handle over a + wide range of sample rate ratios (freq_in/freq_out) from 1:24 to 24:1. + ASRC has two modes of operation. One where ratio can be programmed in SW + and the other where it gets the information from ratio estimator module. + + It supports sample rate conversions in the range of 8 to 192 kHz and + supports 6 streams upto 12 total channels. The input data size can be + 16, 24 and 32 bits. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^asrc@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra186-asrc + - items: + - enum: + - nvidia,tegra234-asrc + - nvidia,tegra194-asrc + - const: nvidia,tegra186-asrc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^ASRC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + ASRC has seven input ports and six output ports. Accordingly ACIF + (Audio Client Interfaces) port nodes are defined to represent the + ASRC inputs (port 0 to 6) and outputs (port 7 to 12). These are + connected to corresponding ports on AHUB (Audio Hub). Additional + input (port 6) is for receiving ratio information from estimator. + + patternProperties: + '^port@[0-6]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF input ports + '^port@[7-9]|1[1-2]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + asrc@2910000 { + compatible = "nvidia,tegra186-asrc"; + reg = <0x2910000 0x2000>; + sound-name-prefix = "ASRC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml new file mode 100644 index 000000000000..73b98b2f3543 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra186-dspk.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra186 DSPK Controller Device Tree Bindings + +description: | + The Digital Speaker Controller (DSPK) can be viewed as a Pulse + Density Modulation (PDM) transmitter that up-samples the input to + the desired sampling rate by interpolation and then converts the + over sampled Pulse Code Modulation (PCM) input to the desired 1-bit + output via Delta Sigma Modulation (DSM). + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^dspk@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra186-dspk + - items: + - enum: + - nvidia,tegra234-dspk + - nvidia,tegra194-dspk + - const: nvidia,tegra186-dspk + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: dspk + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + sound-name-prefix: + pattern: "^DSPK[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DSPK ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DSPK DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + - sound-name-prefix + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra186-clock.h> + + dspk@2905000 { + compatible = "nvidia,tegra186-dspk"; + reg = <0x2905000 0x100>; + clocks = <&bpmp TEGRA186_CLK_DSPK1>; + clock-names = "dspk"; + assigned-clocks = <&bpmp TEGRA186_CLK_DSPK1>; + assigned-clock-parents = <&bpmp TEGRA186_CLK_PLL_A_OUT0>; + assigned-clock-rates = <12288000>; + sound-name-prefix = "DSPK1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt deleted file mode 100644 index dc30c6bfbe95..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt +++ /dev/null @@ -1,30 +0,0 @@ -NVIDIA Tegra 20 I2S controller - -Required properties: -- compatible : "nvidia,tegra20-i2s" -- reg : Should contain I2S registers location and length -- interrupts : Should contain I2S interrupt -- resets : Must contain an entry for each entry in reset-names. - See ../reset/reset.txt for details. -- reset-names : Must include the following entries: - - i2s -- dmas : Must contain an entry for each entry in clock-names. - See ../dma/dma.txt for details. -- dma-names : Must include the following entries: - - rx - - tx -- clocks : Must contain one entry, for the module clock. - See ../clocks/clock-bindings.txt for details. - -Example: - -i2s@70002800 { - compatible = "nvidia,tegra20-i2s"; - reg = <0x70002800 0x200>; - interrupts = < 45 >; - clocks = <&tegra_car 11>; - resets = <&tegra_car 11>; - reset-names = "i2s"; - dmas = <&apbdma 21>, <&apbdma 21>; - dma-names = "rx", "tx"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml new file mode 100644 index 000000000000..68ae124eaf80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 I2S Controller + +description: | + The I2S Controller streams synchronous serial audio data between system + memory and an external audio device. The controller supports the I2S Left + Justified Mode, Right Justified Mode, and DSP mode formats. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-i2s + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: i2s + + interrupts: + maxItems: 1 + + clocks: + minItems: 1 + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = <45>; + clocks = <&tegra_car 11>; + resets = <&tegra_car 11>; + reset-names = "i2s"; + dmas = <&apbdma 21>, <&apbdma 21>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml new file mode 100644 index 000000000000..60a368a132b8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 S/PDIF Controller + +description: | + The S/PDIF controller supports both input and output in serial audio + digital interface format. The input controller can digitally recover + a clock from the received stream. The S/PDIF controller is also used + to generate the embedded audio for HDMI output channel. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 2 + + clock-names: + items: + - const: out + - const: in + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + spdif@70002400 { + compatible = "nvidia,tegra20-spdif"; + reg = <0x70002400 0x200>; + interrupts = <77>; + clocks = <&clk 99>, <&clk 98>; + clock-names = "out", "in"; + resets = <&rst 10>; + dmas = <&apbdma 3>, <&apbdma 3>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml new file mode 100644 index 000000000000..372043edd98f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml @@ -0,0 +1,129 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-admaif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 ADMAIF Device Tree Bindings + +description: | + ADMAIF is the interface between ADMA and AHUB. Each ADMA channel + that sends/receives data to/from AHUB must interface through an + ADMAIF channel. ADMA channel sending data to AHUB pairs with ADMAIF + Tx channel and ADMA channel receiving data from AHUB pairs with + ADMAIF Rx channel. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + $nodename: + pattern: "^admaif@[0-9a-f]*$" + + compatible: + oneOf: + - enum: + - nvidia,tegra210-admaif + - nvidia,tegra186-admaif + - items: + - enum: + - nvidia,tegra234-admaif + - nvidia,tegra194-admaif + - const: nvidia,tegra186-admaif + + reg: + maxItems: 1 + + dmas: true + + dma-names: true + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Contains list of ACIF (Audio CIF) port nodes for ADMAIF channels. + The number of port nodes depends on the number of ADMAIF channels + that SoC may have. These are interfaced with respective ACIF ports + in AHUB (Audio Hub). Each port is capable of data transfers in + both directions. + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +if: + properties: + compatible: + contains: + const: nvidia,tegra210-admaif + +then: + properties: + dmas: + description: + DMA channel specifiers, equally divided for Tx and Rx. + minItems: 1 + maxItems: 20 + dma-names: + items: + pattern: "^[rt]x(10|[1-9])$" + description: + Should be "rx1", "rx2" ... "rx10" for DMA Rx channel + Should be "tx1", "tx2" ... "tx10" for DMA Tx channel + minItems: 1 + maxItems: 20 + +else: + properties: + dmas: + description: + DMA channel specifiers, equally divided for Tx and Rx. + minItems: 1 + maxItems: 40 + dma-names: + items: + pattern: "^[rt]x(1[0-9]|[1-9]|20)$" + description: + Should be "rx1", "rx2" ... "rx20" for DMA Rx channel + Should be "tx1", "tx2" ... "tx20" for DMA Tx channel + minItems: 1 + maxItems: 40 + +required: + - compatible + - reg + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml new file mode 100644 index 000000000000..8d8dc7fb3f0c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-adx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 ADX Device Tree Bindings + +description: | + The Audio Demultiplexer (ADX) block takes an input stream with up to + 16 channels and demultiplexes it into four output streams of up to 16 + channels each. A byte RAM helps to form output frames by any combination + of bytes from the input frame. Its design is identical to that of byte + RAM in the AMX except that the data flow direction is reversed. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^adx@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-adx + - items: + - enum: + - nvidia,tegra234-adx + - nvidia,tegra194-adx + - nvidia,tegra186-adx + - const: nvidia,tegra210-adx + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^ADX[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + ADX has one input and four outputs. Accordingly ACIF (Audio Client + Interface) port nodes are defined to represent ADX input (port 0) + and outputs (ports 1 to 4). These are connected to corresponding + ports on AHUB (Audio Hub). + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF input port + patternProperties: + '^port@[1-4]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + adx@702d3800 { + compatible = "nvidia,tegra210-adx"; + reg = <0x702d3800 0x100>; + sound-name-prefix = "ADX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml new file mode 100644 index 000000000000..6df6f858038c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml @@ -0,0 +1,192 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ahub.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 AHUB Device Tree Bindings + +description: | + The Audio Hub (AHUB) comprises a collection of hardware accelerators + for audio pre-processing, post-processing and a programmable full + crossbar for routing audio data across these accelerators. It has + external interfaces such as I2S, DMIC, DSPK. It interfaces with ADMA + engine through ADMAIF. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + $nodename: + pattern: "^ahub@[0-9a-f]*$" + + compatible: + oneOf: + - enum: + - nvidia,tegra210-ahub + - nvidia,tegra186-ahub + - nvidia,tegra234-ahub + - items: + - const: nvidia,tegra194-ahub + - const: nvidia,tegra186-ahub + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: ahub + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + ranges: true + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Contains list of ACIF (Audio CIF) port nodes for AHUB (Audio Hub). + These are connected to ACIF interfaces of AHUB clients. Thus the + number of port nodes depend on the number of clients that AHUB may + have depending on the SoC revision. + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +patternProperties: + '^i2s@[0-9a-f]+$': + type: object + + '^dmic@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-dmic.yaml# + + '^admaif@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-admaif.yaml# + + '^dspk@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-dspk.yaml# + + '^mvc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mvc.yaml# + + '^sfc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-sfc.yaml# + + '^amx@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-amx.yaml# + + '^adx@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-adx.yaml# + + '^amixer@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mixer.yaml# + + '^asrc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-asrc.yaml# + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + - "#address-cells" + - "#size-cells" + - ranges + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + ahub@702d0800 { + compatible = "nvidia,tegra210-ahub"; + reg = <0x702d0800 0x800>; + clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + clock-names = "ahub"; + assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0x702d0000 0x702d0000 0x0000e400>; + + // All AHUB child nodes below + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + }; + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + reg = <0x702d1000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + sound-name-prefix = "I2S1"; + }; + + dmic@702d4000 { + compatible = "nvidia,tegra210-dmic"; + reg = <0x702d4000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + clock-names = "dmic"; + assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <3072000>; + sound-name-prefix = "DMIC1"; + }; + + // More child nodes to follow + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml new file mode 100644 index 000000000000..f9e4fc6e0c47 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-amx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 AMX Device Tree Bindings + +description: | + The Audio Multiplexer (AMX) block can multiplex up to four input streams + each of which can have maximum 16 channels and generate an output stream + with maximum 16 channels. A byte RAM helps to form an output frame by + any combination of bytes from the input frames. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^amx@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-amx + - items: + - const: nvidia,tegra186-amx + - const: nvidia,tegra210-amx + - const: nvidia,tegra194-amx + - items: + - const: nvidia,tegra234-amx + - const: nvidia,tegra194-amx + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^AMX[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + AMX has four inputs and one output. Accordingly ACIF (Audio Client + Interfaces) port nodes are defined to represent AMX inputs (port 0 + to 3) and output (port 4). These are connected to corresponding + ports on AHUB (Audio Hub). + + patternProperties: + '^port@[0-3]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF input ports + + properties: + port@4: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF output port + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amx@702d3000 { + compatible = "nvidia,tegra210-amx"; + reg = <0x702d3000 0x100>; + sound-name-prefix = "AMX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml new file mode 100644 index 000000000000..bcb496d3ace5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 DMIC Controller Device Tree Bindings + +description: | + The Digital MIC (DMIC) Controller is used to interface with Pulse + Density Modulation (PDM) input devices. It converts PDM signals to + Pulse Coded Modulation (PCM) signals. DMIC can be viewed as a PDM + receiver. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^dmic@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-dmic + - items: + - enum: + - nvidia,tegra234-dmic + - nvidia,tegra194-dmic + - nvidia,tegra186-dmic + - const: nvidia,tegra210-dmic + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: dmic + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + sound-name-prefix: + pattern: "^DMIC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC DAP (Digital Audio Port) interface which can be connected + to external audio codec for capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + dmic@702d4000 { + compatible = "nvidia,tegra210-dmic"; + reg = <0x702d4000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + clock-names = "dmic"; + assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <3072000>; + sound-name-prefix = "DMIC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml new file mode 100644 index 000000000000..6188f561f878 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 I2S Controller Device Tree Bindings + +description: | + The Inter-IC Sound (I2S) controller implements full-duplex, + bi-directional and single direction point-to-point serial + interfaces. It can interface with I2S compatible devices. + I2S controller can operate both in master and slave mode. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^i2s@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-i2s + - items: + - enum: + - nvidia,tegra234-i2s + - nvidia,tegra194-i2s + - nvidia,tegra186-i2s + - const: nvidia,tegra210-i2s + + reg: + maxItems: 1 + + clocks: + minItems: 1 + items: + - description: I2S bit clock + - description: + Sync input clock, which can act as clock source to other I/O + modules in AHUB. The Tegra I2S driver sets this clock rate as + per bit clock rate. I/O module which wants to use this clock + as source, can mention this clock as parent in the DT bindings. + This is an optional clock entry, since it is only required when + some other I/O wants to reference from a particular I2Sx + instance. + + clock-names: + minItems: 1 + items: + - const: i2s + - const: sync_input + + assigned-clocks: + minItems: 1 + maxItems: 2 + + assigned-clock-parents: + minItems: 1 + maxItems: 2 + + assigned-clock-rates: + minItems: 1 + maxItems: 2 + + sound-name-prefix: + pattern: "^I2S[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback or capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + reg = <0x702d1000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + sound-name-prefix = "I2S1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml new file mode 100644 index 000000000000..ee1e1d2da79a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mixer.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 Mixer Device Tree Bindings + +description: | + The Mixer supports mixing of up to ten 7.1 audio input streams and + generate five outputs (each of which can be any combination of the + ten input streams). + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^amixer@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-amixer + - items: + - enum: + - nvidia,tegra234-amixer + - nvidia,tegra194-amixer + - nvidia,tegra186-amixer + - const: nvidia,tegra210-amixer + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^MIXER[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Mixer has ten inputs and five outputs. Accordingly ACIF (Audio + Client Interfaces) port nodes are defined to represent Mixer + inputs (port 0 to 9) and outputs (port 10 to 14). These are + connected to corresponding ports on AHUB (Audio Hub). + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF input ports + '^port@[10-14]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amixer@702dbb00 { + compatible = "nvidia,tegra210-amixer"; + reg = <0x702dbb00 0x800>; + sound-name-prefix = "MIXER1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml new file mode 100644 index 000000000000..c9888c553e78 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mvc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 MVC Device Tree Bindings + +description: | + The Master Volume Control (MVC) provides gain or attenuation to a digital + signal path. It can be used in input or output signal path for per-stream + volume control or it can be used as master volume control. The MVC block + has one input and one output. The input digital stream can be mono or + multi-channel (up to 7.1 channels) stream. An independent mute control is + also included in the MVC block. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^mvc@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-mvc + - items: + - enum: + - nvidia,tegra234-mvc + - nvidia,tegra194-mvc + - nvidia,tegra186-mvc + - const: nvidia,tegra210-mvc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^MVC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + mvc@702da000 { + compatible = "nvidia,tegra210-mvc"; + reg = <0x702da000 0x200>; + sound-name-prefix = "MVC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml new file mode 100644 index 000000000000..8579306fc56f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-sfc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 SFC Device Tree Bindings + +description: | + The Sampling Frequency Converter (SFC) converts the sampling frequency + of the input signal from one frequency to another. It supports sampling + frequency conversions of streams of up to two channels (stereo). + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^sfc@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-sfc + - items: + - enum: + - nvidia,tegra234-sfc + - nvidia,tegra194-sfc + - nvidia,tegra186-sfc + - const: nvidia,tegra210-sfc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^SFC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + SFC ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + SFC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + sfc@702d2000 { + compatible = "nvidia,tegra210-sfc"; + reg = <0x702d2000 0x200>; + sound-name-prefix = "SFC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt deleted file mode 100644 index 21cd310963b1..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt +++ /dev/null @@ -1,35 +0,0 @@ -NVIDIA Tegra30 HDA controller - -Required properties: -- compatible : For Tegra30, must contain "nvidia,tegra30-hda". Otherwise, - must contain '"nvidia,<chip>-hda", "nvidia,tegra30-hda"', where <chip> is - tegra114, tegra124, or tegra132. -- reg : Should contain the HDA registers location and length. -- interrupts : The interrupt from the HDA controller. -- clocks : Must contain an entry for each required entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x -- resets : Must contain an entry for each entry in reset-names. - 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 { - compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; - reg = <0x0 0x70030000 0x0 0x10000>; - interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&tegra_car TEGRA124_CLK_HDA>, - <&tegra_car TEGRA124_CLK_HDA2HDMI>, - <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; - clock-names = "hda", "hda2hdmi", "hda2codec_2x"; - resets = <&tegra_car 125>, /* hda */ - <&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/nvidia,tegra30-hda.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml new file mode 100644 index 000000000000..12c31b4b99e1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra30-hda.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra HDA controller + +description: | + The High Definition Audio (HDA) block provides a serial interface to + audio codec. It supports multiple input and output streams. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + $nodename: + pattern: "^hda@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra30-hda + - items: + - enum: + - nvidia,tegra234-hda + - nvidia,tegra194-hda + - nvidia,tegra186-hda + - nvidia,tegra210-hda + - nvidia,tegra124-hda + - const: nvidia,tegra30-hda + - items: + - const: nvidia,tegra132-hda + - const: nvidia,tegra124-hda + - const: nvidia,tegra30-hda + + reg: + maxItems: 1 + + interrupts: + description: The interrupt from the HDA controller + maxItems: 1 + + clocks: + minItems: 2 + maxItems: 3 + + clock-names: + minItems: 2 + items: + - const: hda + - const: hda2hdmi + - const: hda2codec_2x + + resets: + minItems: 2 + maxItems: 3 + + reset-names: + minItems: 2 + items: + - const: hda + - const: hda2hdmi + - const: hda2codec_2x + + power-domains: + maxItems: 1 + + interconnects: + maxItems: 2 + + interconnect-names: + items: + - const: dma-mem + - const: write + + iommus: + maxItems: 1 + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: | + The user-visible name of this sound complex. If this property is + not specified then boards can use default name provided in hda driver. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra124-car-common.h> + #include<dt-bindings/interrupt-controller/arm-gic.h> + + hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hda2codec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>, /* hda2hdmi */ + <&tegra_car 111>; /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hda2codec_2x"; + nvidia,model = "jetson-tk1-hda"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml new file mode 100644 index 000000000000..b9b1dba40856 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,tfa989x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP/Goodix TFA989X (TFA1) Audio Amplifiers + +maintainers: + - Stephan Gerhold <stephan@gerhold.net> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + enum: + - nxp,tfa9895 + - nxp,tfa9897 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + rcv-gpios: + description: optional GPIO to be asserted when receiver mode is enabled. + + sound-name-prefix: true + + vddd-supply: + description: regulator phandle for the VDDD power supply. + +if: + not: + properties: + compatible: + const: nxp,tfa9897 +then: + properties: + rcv-gpios: false + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@34 { + compatible = "nxp,tfa9895"; + reg = <0x34>; + sound-name-prefix = "Speaker Left"; + #sound-dai-cells = <0>; + }; + audio-codec@36 { + compatible = "nxp,tfa9895"; + reg = <0x36>; + sound-name-prefix = "Speaker Right"; + #sound-dai-cells = <0>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + speaker_codec_top: audio-codec@34 { + compatible = "nxp,tfa9897"; + reg = <0x34>; + vddd-supply = <&pm8916_l6>; + rcv-gpios = <&msmgpio 50 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&speaker_top_default>; + sound-name-prefix = "Speaker Top"; + #sound-dai-cells = <0>; + }; + + speaker_codec_bottom: audio-codec@36 { + compatible = "nxp,tfa9897"; + reg = <0x36>; + vddd-supply = <&pm8916_l6>; + rcv-gpios = <&msmgpio 111 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&speaker_bottom_default>; + sound-name-prefix = "Speaker Bottom"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt deleted file mode 100644 index 84b28dbe9f15..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt +++ /dev/null @@ -1,89 +0,0 @@ -* Qualcomm Technologies APQ8016 SBC ASoC machine driver - -This node models the Qualcomm Technologies APQ8016 SBC ASoC machine driver - -Required properties: - -- compatible : "qcom,apq8016-sbc-sndcard" - -- pinctrl-N : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. -- pinctrl-names : Must contain a "default" entry. -- reg : Must contain an address for each entry in reg-names. -- reg-names : A list which must include the following entries: - * "mic-iomux" - * "spkr-iomux" -- qcom,model : Name of the sound card. - -- qcom,audio-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 could be power supplies, MicBias - of msm8x16_wcd codec and the jacks on the board: - - Power supplies: - * MIC BIAS External1 - * MIC BIAS External2 - * MIC BIAS Internal1 - * MIC BIAS Internal2 - - Board connectors: - * Headset Mic - * Secondary Mic - * DMIC - * Ext Spk - -Dai-link subnode properties and subnodes: - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Required CPU/CODEC subnodes properties: - --link-name : Name of the dai link. --sound-dai : phandle/s and port of CPU/CODEC - -Example: - -sound: sound { - compatible = "qcom,apq8016-sbc-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - qcom,model = "DB410c"; - - qcom,audio-routing = - "MIC BIAS External1", "Handset Mic", - "MIC BIAS Internal2", "Headset Mic", - "MIC BIAS External1", "Secondary Mic", - "AMIC1", "MIC BIAS External1", - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS External1", - "DMIC1", "MIC BIAS Internal1", - "MIC BIAS Internal1", "Digital Mic1", - "DMIC2", "MIC BIAS Internal1", - "MIC BIAS Internal1", "Digital Mic2"; - - /* I2S - Internal codec */ - internal-dai-link@0 { - cpu { /* PRIMARY */ - sound-dai = <&lpass MI2S_PRIMARY>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - - /* External Primary or External Secondary -ADV7533 HDMI */ - external-dai-link@0 { - link-name = "ADV7533"; - cpu { /* QUAT */ - sound-dai = <&lpass MI2S_QUATERNARY>; - }; - codec { - sound-dai = <&adv_bridge 0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt index c814e867850f..e1b9fa8a5bf8 100644 --- a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt +++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt @@ -55,6 +55,14 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. Value type: <stringlist> Definition: The user-visible name of this sound card. +- aux-devs + Usage: optional + Value type: <array of phandles> + Definition: A list of phandles for auxiliary devices (e.g. analog + amplifiers) that do not appear directly within the DAI + links. Should be connected to another audio component + using "audio-routing". + = dailinks Each subnode of sndcard represents either a dailink, and subnodes of each dailinks would be cpu/codec/platform dais. diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt deleted file mode 100644 index 21c648328be9..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ /dev/null @@ -1,54 +0,0 @@ -* Qualcomm Technologies LPASS CPU DAI - -This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS). - -Required properties: - -- compatible : "qcom,lpass-cpu" or "qcom,apq8016-lpass-cpu" -- clocks : Must contain an entry for each entry in clock-names. -- clock-names : A list which must include the following entries: - * "ahbix-clk" - * "mi2s-osr-clk" - * "mi2s-bit-clk" - : required clocks for "qcom,lpass-cpu-apq8016" - * "ahbix-clk" - * "mi2s-bit-clk0" - * "mi2s-bit-clk1" - * "mi2s-bit-clk2" - * "mi2s-bit-clk3" - * "pcnoc-mport-clk" - * "pcnoc-sway-clk" - -- interrupts : Must contain an entry for each entry in - interrupt-names. -- interrupt-names : A list which must include the following entries: - * "lpass-irq-lpaif" -- pinctrl-N : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. -- pinctrl-names : Must contain a "default" entry. -- reg : Must contain an address for each entry in reg-names. -- reg-names : A list which must include the following entries: - * "lpass-lpaif" - - - -Optional properties: - -- qcom,adsp : Phandle for the audio DSP node - -Example: - -lpass@28100000 { - compatible = "qcom,lpass-cpu"; - clocks = <&lcc AHBIX_CLK>, <&lcc MI2S_OSR_CLK>, <&lcc MI2S_BIT_CLK>; - clock-names = "ahbix-clk", "mi2s-osr-clk", "mi2s-bit-clk"; - interrupts = <0 85 1>; - interrupt-names = "lpass-irq-lpaif"; - pinctrl-names = "default", "idle"; - pinctrl-0 = <&mi2s_default>; - pinctrl-1 = <&mi2s_idle>; - reg = <0x28100000 0x10000>; - reg-names = "lpass-lpaif"; - qcom,adsp = <&adsp>; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml new file mode 100644 index 000000000000..e9a533080b32 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -0,0 +1,293 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-cpu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. LPASS CPU dai driver bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + - Rohit kumar <rohitkr@codeaurora.org> + +description: | + Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist + of MI2S interface for audio data transfer on external codecs. LPASS cpu driver + is a module to configure Low-Power Audio Interface(LPAIF) core registers + across different IP versions. + +properties: + compatible: + enum: + - qcom,lpass-cpu + - qcom,apq8016-lpass-cpu + - qcom,sc7180-lpass-cpu + - qcom,sc7280-lpass-cpu + + reg: + minItems: 2 + maxItems: 6 + description: LPAIF core registers + + reg-names: + minItems: 2 + maxItems: 6 + + clocks: + minItems: 3 + maxItems: 7 + + clock-names: + minItems: 1 + maxItems: 10 + + interrupts: + minItems: 2 + maxItems: 4 + description: LPAIF DMA buffer interrupt + + interrupt-names: + minItems: 2 + maxItems: 4 + + qcom,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle for the audio DSP node + + iommus: + minItems: 2 + maxItems: 3 + description: Phandle to apps_smmu node with sid mask + + power-domains: + maxItems: 1 + + power-domain-names: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]$": + type: object + description: | + LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node + depends on the specific driver providing the functionality and + properties. + properties: + reg: + maxItems: 1 + description: Must be one of the DAI ID + + qcom,playback-sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: list of MI2S data lines for playback + + qcom,capture-sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: list of MI2S data lines for capture + + required: + - reg + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - interrupts + - interrupt-names + - '#sound-dai-cells' + +additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: qcom,lpass-cpu + + then: + properties: + clock-names: + items: + - const: ahbix-clk + - const: mi2s-osr-clk + - const: mi2s-bit-clk + + - if: + properties: + compatible: + contains: + const: qcom,apq8016-lpass-cpu + + then: + properties: + clock-names: + items: + - const: ahbix-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + - const: mi2s-bit-clk2 + - const: mi2s-bit-clk3 + - const: pcnoc-mport-clk + - const: pcnoc-sway-clk + + - if: + properties: + compatible: + contains: + const: qcom,sc7180-lpass-cpu + + then: + properties: + clock-names: + oneOf: + - items: #for I2S + - const: pcnoc-sway-clk + - const: audio-core + - const: mclk0 + - const: pcnoc-mport-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + - items: #for HDMI + - const: pcnoc-sway-clk + - const: audio-core + - const: pcnoc-mport-clk + reg-names: + anyOf: + - items: #for I2S + - const: lpass-lpaif + - items: #for I2S and HDMI + - const: lpass-hdmiif + - const: lpass-lpaif + interrupt-names: + anyOf: + - items: #for I2S + - const: lpass-irq-lpaif + - items: #for I2S and HDMI + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + required: + - iommus + - power-domains + + - if: + properties: + compatible: + contains: + const: qcom,sc7280-lpass-cpu + + then: + properties: + clock-names: + oneOf: + - items: #for I2S + - const: aon_cc_audio_hm_h + - const: audio_cc_ext_mclk0 + - const: core_cc_sysnoc_mport_core + - const: core_cc_ext_if0_ibit + - const: core_cc_ext_if1_ibit + - items: #for Soundwire + - const: aon_cc_audio_hm_h + - const: audio_cc_codec_mem + - const: audio_cc_codec_mem0 + - const: audio_cc_codec_mem1 + - const: audio_cc_codec_mem2 + - const: aon_cc_va_mem0 + - items: #for HDMI + - const: core_cc_sysnoc_mport_core + + reg-names: + anyOf: + - items: #for I2S + - const: lpass-lpaif + - items: #for I2S and HDMI + - const: lpass-hdmiif + - const: lpass-lpaif + - items: #for I2S, soundwire and HDMI + - const: lpass-hdmiif + - const: lpass-lpaif + - const: lpass-rxtx-cdc-dma-lpm + - const: lpass-rxtx-lpaif + - const: lpass-va-lpaif + - const: lpass-va-cdc-dma-lpm + interrupt-names: + anyOf: + - items: #for I2S + - const: lpass-irq-lpaif + - items: #for I2S and HDMI + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - items: #for I2S, soundwire and HDMI + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - const: lpass-irq-vaif + - const: lpass-irq-rxtxif + power-domain-names: + allOf: + - items: + - const: lcx + + required: + - iommus + - power-domains + +examples: + - | + #include <dt-bindings/sound/sc7180-lpass.h> + + soc { + #address-cells = <2>; + #size-cells = <2>; + lpass@62d80000 { + compatible = "qcom,sc7180-lpass-cpu"; + + reg = <0 0x62d87000 0 0x68000>, + <0 0x62f00000 0 0x29000>; + reg-names = "lpass-hdmiif", + "lpass-lpaif"; + iommus = <&apps_smmu 0x1020 0>, + <&apps_smmu 0x1032 0>; + power-domains = <&lpass_hm 0>; + + clocks = <&gcc 131>, + <&lpasscorecc 6>, + <&lpasscorecc 7>, + <&lpasscorecc 10>, + <&lpasscorecc 8>, + <&lpasscorecc 9>; + + clock-names = "pcnoc-sway-clk", "audio-core", + "mclk0", "pcnoc-mport-clk", + "mi2s-bit-clk0", "mi2s-bit-clk1"; + + interrupts = <0 160 1>, + <0 268 1>; + interrupt-names = "lpass-irq-lpaif", + "lpass-irq-hdmi"; + #sound-dai-cells = <1>; + + #address-cells = <1>; + #size-cells = <0>; + /* Optional to set different MI2S SD lines */ + dai-link@0 { + reg = <MI2S_PRIMARY>; + qcom,playback-sd-lines = <1>; + qcom,capture-sd-lines = <0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml new file mode 100644 index 000000000000..a6905bcf89d2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-rx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) RX Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 5 + + clock-names: + oneOf: + - items: #for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: #for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + clock-output-names: + items: + - const: mclk + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3200000 { + compatible = "qcom,sm8250-lpass-rx-macro"; + reg = <0x3200000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 0>, + <&audiocc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml new file mode 100644 index 000000000000..324595a62ae8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-tx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) TX Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 5 + + clock-names: + oneOf: + - items: #for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: #for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + clock-output-names: + items: + - const: mclk + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3220000 { + compatible = "qcom,sm8250-lpass-tx-macro"; + reg = <0x3220000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&aoncc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + qcom,dmic-sample-rate = <600000>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml new file mode 100644 index 000000000000..7b4cc84eda8c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-va-macro + - qcom,sm8250-lpass-va-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 3 + + clock-names: + oneOf: + - items: #for ADSP based platforms + - const: mclk + - const: core + - const: dcodec + - items: #for ADSP bypass based platforms + - const: mclk + + clock-output-names: + items: + - const: fsgen + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + + vdd-micb-supply: + description: phandle to voltage regulator of MIC Bias + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3370000 { + compatible = "qcom,sm8250-lpass-va-macro"; + reg = <0x3370000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; + clock-names = "mclk", "core", "dcodec"; + clock-output-names = "fsgen"; + qcom,dmic-sample-rate = <600000>; + vdd-micb-supply = <&vreg_s4a_1p8>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml new file mode 100644 index 000000000000..13cdb8a10687 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-wsa-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 5 + + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + clock-output-names: + items: + - const: mclk + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + + vdd-micb-supply: + description: phandle to voltage regulator of MIC Bias + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3240000 { + compatible = "qcom,sm8250-lpass-wsa-macro"; + reg = <0x3240000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 1>, + <&audiocc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt index bbae426cdfb1..15c353a20de8 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt @@ -29,7 +29,7 @@ used by the apr service device. Definition: Must be 0 = EXAMPLE -q6adm@8 { +apr-service@8 { compatible = "qcom,q6adm"; reg = <APR_SVC_ADM>; q6routing: routing { diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index d74888b9f1bb..bc6b5f1fe4f1 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -12,167 +12,9 @@ used by all apr services. Must contain the following properties. from DSP. example "qcom,q6afe" -= AFE DAIs (Digial Audio Interface) -"dais" subnode of the AFE node. It represents afe dais, each afe dai is a -subnode of "dais" representing board specific dai setup. -"dais" node should have following properties followed by dai children. - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6afe-dais" - -- #sound-dai-cells - Usage: required - Value type: <u32> - Definition: Must be 1 - -- #address-cells - Usage: required - Value type: <u32> - Definition: Must be 1 - -- #size-cells - Usage: required - Value type: <u32> - Definition: Must be 0 - -== AFE 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 - -- qcom,sd-lines - Usage: required for mi2s interface - Value type: <prop-encoded-array> - Definition: Must be list of serial data lines used by this dai. - should be one or more of the 0-3 sd lines. - - - qcom,tdm-sync-mode: - Usage: required for tdm interface - Value type: <prop-encoded-array> - Definition: Synchronization mode. - 0 - Short sync bit mode - 1 - Long sync mode - 2 - Short sync slot mode - - - qcom,tdm-sync-src: - Usage: required for tdm interface - Value type: <prop-encoded-array> - Definition: Synchronization source. - 0 - External source - 1 - Internal source - - - qcom,tdm-data-out: - Usage: required for tdm interface - Value type: <prop-encoded-array> - Definition: Data out signal to drive with other masters. - 0 - Disable - 1 - Enable - - - qcom,tdm-invert-sync: - Usage: required for tdm interface - Value type: <prop-encoded-array> - Definition: Invert the sync. - 0 - Normal - 1 - Invert - - - qcom,tdm-data-delay: - Usage: required for tdm interface - Value type: <prop-encoded-array> - Definition: Number of bit clock to delay data - with respect to sync edge. - 0 - 0 bit clock cycle - 1 - 1 bit clock cycle - 2 - 2 bit clock cycle - - - qcom,tdm-data-align: - Usage: required for tdm interface - Value type: <prop-encoded-array> - Definition: Indicate how data is packed - within the slot. For example, 32 slot width in case of - sample bit width is 24. - 0 - MSB - 1 - LSB - = EXAMPLE -q6afe@4 { +apr-service@4 { compatible = "qcom,q6afe"; reg = <APR_SVC_AFE>; - - dais { - compatible = "qcom,q6afe-dais"; - #sound-dai-cells = <1>; - #address-cells = <1>; - #size-cells = <0>; - - hdmi@1 { - reg = <1>; - }; - - tdm@24 { - reg = <24>; - qcom,tdm-sync-mode = <1>: - qcom,tdm-sync-src = <1>; - qcom,tdm-data-out = <0>; - qcom,tdm-invert-sync = <1>; - qcom,tdm-data-delay = <1>; - qcom,tdm-data-align = <0>; - - }; - - tdm@25 { - reg = <25>; - qcom,tdm-sync-mode = <1>: - qcom,tdm-sync-src = <1>; - qcom,tdm-data-out = <0>; - qcom,tdm-invert-sync = <1>; - qcom,tdm-data-delay <1>: - qcom,tdm-data-align = <0>; - }; - - prim-mi2s-rx@16 { - reg = <16>; - qcom,sd-lines = <0 2>; - }; - - prim-mi2s-tx@17 { - reg = <17>; - qcom,sd-lines = <1>; - }; - - sec-mi2s-rx@18 { - reg = <18>; - qcom,sd-lines = <0 3>; - }; - - sec-mi2s-tx@19 { - reg = <19>; - qcom,sd-lines = <1>; - }; - - tert-mi2s-rx@20 { - reg = <20>; - qcom,sd-lines = <1 3>; - }; - - tert-mi2s-tx@21 { - reg = <21>; - qcom,sd-lines = <0>; - }; - - quat-mi2s-rx@22 { - reg = <22>; - qcom,sd-lines = <0>; - }; - - quat-mi2s-tx@23 { - reg = <23>; - qcom,sd-lines = <1>; - }; - }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml new file mode 100644 index 000000000000..5d972784321d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml @@ -0,0 +1,53 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/qcom,q6apm-dai.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Qualcomm Audio Process Manager Digital Audio Interfaces binding + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm APM DAIs in DSP + +properties: + compatible: + const: qcom,q6apm-dais + + reg: + maxItems: 1 + + iommus: + maxItems: 1 + +required: + - compatible + - iommus + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,gpr.h> + gpr { + compatible = "qcom,gpr"; + #address-cells = <1>; + #size-cells = <0>; + qcom,domain = <GPR_DOMAIN_ID_ADSP>; + service@1 { + compatible = "qcom,q6apm"; + reg = <1>; + + #address-cells = <1>; + #size-cells = <0>; + + apm-dai@1 { + compatible = "qcom,q6apm-dais"; + iommus = <&apps_smmu 0x1801 0x0>; + reg = <1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 9f5378c51686..0d0075125243 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -14,7 +14,7 @@ used by the apr service device. from DSP. example "qcom,q6asm-v2.0" -= ASM DAIs (Digial Audio Interface) += ASM DAIs (Digital Audio Interface) "dais" subnode of the ASM node represents dai specific configuration - compatible: @@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties. 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) + Q6ASM_DAI_TX_RX (0) for both tx and rx + Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + Q6ASM_DAI_RX (2) for only rx (Playback/Decode) - is-compress-dai: Usage: Required for Compress offload dais @@ -50,16 +50,20 @@ configuration of each dai. Must contain the following properties. = EXAMPLE +#include <dt-bindings/sound/qcom,q6asm.h> -q6asm@7 { +apr-service@7 { compatible = "qcom,q6asm"; reg = <APR_SVC_ASM>; q6asmdai: dais { compatible = "qcom,q6asm-dais"; + #address-cells = <1>; + #size-cells = <0>; #sound-dai-cells = <1>; - mm@0 { + + dai@0 { reg = <0>; - direction = <2>; + direction = <Q6ASM_DAI_RX>; is-compress-dai; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt index 7f36ff8bec18..5cd4cc9b1fde 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6core.txt @@ -15,7 +15,7 @@ used by the apr service device. example "qcom,q6core-v2.0" = EXAMPLE -q6core@3 { +apr-service@3 { compatible = "qcom,q6core"; reg = <APR_SVC_ADSP_CORE>; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml new file mode 100644 index 000000000000..f83f00737a2f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-clocks.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Qualcomm DSP LPASS Clock Controller binding + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP Clock Controller + +properties: + compatible: + enum: + - qcom,q6afe-clocks + - qcom,q6prm-lpass-clocks + + reg: + maxItems: 1 + + '#clock-cells': + const: 2 + description: + Clock Id is followed by clock coupling attributes. + 1 = for no coupled clock + 2 = for dividend of the coupled clock + 3 = for divisor of the coupled clock + 4 = for inverted and no couple clock + +required: + - compatible + - reg + - "#clock-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6afe.h> + apr { + #address-cells = <1>; + #size-cells = <0>; + apr-service@4 { + reg = <APR_SVC_AFE>; + #address-cells = <1>; + #size-cells = <0>; + clock-controller@2 { + compatible = "qcom,q6afe-clocks"; + reg = <2>; + #clock-cells = <2>; + }; + }; + }; + + - | + #include <dt-bindings/soc/qcom,gpr.h> + gpr { + compatible = "qcom,gpr"; + qcom,domain = <GPR_DOMAIN_ID_ADSP>; + #address-cells = <1>; + #size-cells = <0>; + service@2 { + reg = <GPR_PRM_MODULE_IID>; + compatible = "qcom,q6prm"; + #address-cells = <1>; + #size-cells = <0>; + clock-controller@2 { + compatible = "qcom,q6prm-lpass-clocks"; + reg = <2>; + #clock-cells = <2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml new file mode 100644 index 000000000000..dc7fba7b92d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-ports.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Qualcomm DSP LPASS(Low Power Audio SubSystem) Audio Ports binding + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP LPASS Audio ports + +properties: + compatible: + enum: + - qcom,q6afe-dais + - qcom,q6apm-lpass-dais + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +#Digital Audio Interfaces +patternProperties: + '^dai@[0-9]+$': + type: object + description: + Q6DSP Digital Audio Interfaces. + + properties: + reg: + description: + Digital Audio Interface ID + + qcom,sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + List of serial data lines used by this dai.should be one or more of the 0-3 sd lines. + minItems: 1 + maxItems: 4 + uniqueItems: true + items: + minimum: 0 + maximum: 3 + + qcom,tdm-sync-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: + TDM Synchronization mode + 0 = Short sync bit mode + 1 = Long sync mode + 2 = Short sync slot mode + + qcom,tdm-sync-src: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + TDM Synchronization source + 0 = External source + 1 = Internal source + + qcom,tdm-data-out: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + TDM Data out signal to drive with other masters + 0 = Disable + 1 = Enable + + qcom,tdm-invert-sync: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + TDM Invert the sync + 0 = Normal + 1 = Invert + + qcom,tdm-data-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: + TDM Number of bit clock to delay data + 0 = 0 bit clock cycle + 1 = 1 bit clock cycle + 2 = 2 bit clock cycle + + qcom,tdm-data-align: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + Indicate how data is packed within the slot. For example, 32 slot + width in case of sample bit width is 24TDM Invert the sync. + 0 = MSB + 1 = LSB + + required: + - reg + + allOf: + - if: + properties: + reg: + contains: + # TDM DAI ID range from PRIMARY_TDM_RX_0 - QUINARY_TDM_TX_7 + items: + minimum: 24 + maximum: 103 + then: + required: + - qcom,tdm-sync-mode + - qcom,tdm-sync-src + - qcom,tdm-data-out + - qcom,tdm-invert-sync + - qcom,tdm-data-delay + - qcom,tdm-data-align + + - if: + properties: + reg: + contains: + # MI2S DAI ID range PRIMARY_MI2S_RX - QUATERNARY_MI2S_TX and + # QUINARY_MI2S_RX - QUINARY_MI2S_TX + items: + oneOf: + - minimum: 16 + maximum: 23 + - minimum: 127 + maximum: 128 + then: + required: + - qcom,sd-lines + + additionalProperties: false + +required: + - compatible + - reg + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6afe.h> + apr { + #address-cells = <1>; + #size-cells = <0>; + apr-service@4 { + reg = <APR_SVC_AFE>; + #address-cells = <1>; + #size-cells = <0>; + q6afedai@1 { + compatible = "qcom,q6afe-dais"; + reg = <1>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@22 { + reg = <QUATERNARY_MI2S_RX>; + qcom,sd-lines = <0 1 2 3>; + }; + }; + }; + }; + - | + #include <dt-bindings/soc/qcom,gpr.h> + gpr { + compatible = "qcom,gpr"; + #address-cells = <1>; + #size-cells = <0>; + qcom,domain = <GPR_DOMAIN_ID_ADSP>; + service@1 { + compatible = "qcom,q6apm"; + reg = <GPR_APM_MODULE_IID>; + #address-cells = <1>; + #size-cells = <0>; + q6apmdai@1 { + compatible = "qcom,q6apm-lpass-dais"; + reg = <1>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@22 { + reg = <QUATERNARY_MI2S_RX>; + qcom,sd-lines = <0 1 2 3>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt index 408c4837e6d5..de4c604641da 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt @@ -5,7 +5,10 @@ This binding describes the SDM845 sound card, which uses qdsp for audio. - compatible: Usage: required Value type: <stringlist> - Definition: must be "qcom,sdm845-sndcard" + Definition: must be one of this + "qcom,sdm845-sndcard" + "qcom,db845c-sndcard" + "lenovo,yoga-c630-sndcard" - audio-routing: Usage: Optional @@ -21,6 +24,14 @@ This binding describes the SDM845 sound card, which uses qdsp for audio. Value type: <stringlist> Definition: The user-visible name of this sound card. +- aux-devs + Usage: optional + Value type: <array of phandles> + Definition: A list of phandles for auxiliary devices (e.g. analog + amplifiers) that do not appear directly within the DAI + links. Should be connected to another audio component + using "audio-routing". + = dailinks Each subnode of sndcard represents either a dailink, and subnodes of each dailinks would be cpu/codec/platform dais. diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml new file mode 100644 index 000000000000..4ecd4080bb96 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -0,0 +1,303 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,sm8250.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. ASoC sound card drivers + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + This bindings describes Qualcomm SoC based sound cards + which uses LPASS internal codec for audio. + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + - qcom,sm8250-sndcard + - qcom,qrb5165-rb5-sndcard + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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 could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + + pin-switches: + description: List of widget names for which pin switches should be created. + $ref: /schemas/types.yaml#/definitions/string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + # Only valid for some compatibles (see allOf if below) + reg: true + reg-names: true + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: + minItems: 1 + maxItems: 4 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - model + +allOf: + - if: + properties: + compatible: + contains: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + then: + properties: + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + required: + - compatible + - model + - reg + - reg-names + else: + properties: + reg: false + reg-names: false + +additionalProperties: false + +examples: + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound { + compatible = "qcom,qrb5165-rb5-sndcard"; + model = "Qualcomm-qrb5165-RB5-WSA8815-Speakers-DMIC0"; + audio-routing = "SpkrLeft IN", "WSA_SPK1 OUT", + "SpkrRight IN", "WSA_SPK2 OUT", + "VA DMIC0", "vdd-micb", + "VA DMIC1", "vdd-micb"; + + mm1-dai-link { + link-name = "MultiMedia0"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + mm2-dai-link { + link-name = "MultiMedia2"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA2>; + }; + }; + + mm3-dai-link { + link-name = "MultiMedia3"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA3>; + }; + }; + + hdmi-dai-link { + link-name = "HDMI Playback"; + cpu { + sound-dai = <&q6afedai TERTIARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + + codec { + sound-dai = <<9611_codec 0>; + }; + }; + + wsa-dai-link { + link-name = "WSA Playback"; + cpu { + sound-dai = <&q6afedai WSA_CODEC_DMA_RX_0>; + }; + + platform { + sound-dai = <&q6routing>; + }; + + codec { + sound-dai = <&left_spkr>, <&right_spkr>, <&swr0 0>, <&wsamacro>; + }; + }; + + va-dai-link { + link-name = "VA Capture"; + cpu { + sound-dai = <&q6afedai VA_CODEC_DMA_TX_0>; + }; + + platform { + sound-dai = <&q6routing>; + }; + + codec { + sound-dai = <&vamacro 0>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml new file mode 100644 index 000000000000..9b225dbf8b79 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -0,0 +1,206 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd934x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WCD9340/WCD9341 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9340/WCD9341 Codec is a standalone Hi-Fi audio codec IC. + It has in-built Soundwire controller, pin controller, interrupt mux and + supports both I2S/I2C and SLIMbus audio interfaces. + +properties: + compatible: + const: slim217,250 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + reset-gpios: + description: GPIO spec for reset line to use + maxItems: 1 + + slim-ifc-dev: true + + clocks: + maxItems: 1 + + clock-names: + const: extclk + + vdd-buck-supply: + description: A reference to the 1.8V buck supply + + vdd-buck-sido-supply: + description: A reference to the 1.8V SIDO buck supply + + vdd-rx-supply: + description: A reference to the 1.8V rx supply + + vdd-tx-supply: + description: A reference to the 1.8V tx supply + + vdd-vbat-supply: + description: A reference to the vbat supply + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-micbias-supply: + description: A reference to the micbias supply + + qcom,micbias1-microvolt: + description: micbias1 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias2-microvolt: + description: micbias2 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias3-microvolt: + description: micbias3 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias4-microvolt: + description: micbias4 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,hphl-jack-type-normally-closed: + description: Indicates that HPHL jack switch type is normally closed + type: boolean + + qcom,ground-jack-type-normally-closed: + description: Indicates that Headset Ground switch type is normally closed + type: boolean + + qcom,mbhc-headset-vthreshold-microvolt: + description: Voltage threshold value for headset detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-headphone-vthreshold-microvolt: + description: Voltage threshold value for headphone detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-buttons-vthreshold-microvolt: + description: + Array of 8 Voltage threshold values corresponding to headset + button0 - button7 + minItems: 8 + maxItems: 8 + + clock-output-names: + const: mclk + + clock-frequency: + description: Clock frequency of output clk in Hz + + interrupt-controller: true + + '#interrupt-cells': + const: 1 + + '#clock-cells': + const: 0 + + '#sound-dai-cells': + const: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + gpio@42: + type: object + $ref: ../gpio/qcom,wcd934x-gpio.yaml# + +patternProperties: + "^.*@[0-9a-f]+$": + type: object + description: | + WCD934x subnode for each slave devices. Bindings of each subnodes + depends on the specific driver providing the functionality and + documented in their respective bindings. + + properties: + reg: + maxItems: 1 + + required: + - reg + +required: + - compatible + - reg + - reset-gpios + - slim-ifc-dev + - interrupts + - interrupt-controller + - clock-frequency + - clock-output-names + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#interrupt-cells" + - "#clock-cells" + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + codec@1,0{ + compatible = "slim217,250"; + reg = <1 0>; + reset-gpios = <&tlmm 64 0>; + slim-ifc-dev = <&wcd9340_ifd>; + #sound-dai-cells = <1>; + interrupt-parent = <&tlmm>; + interrupts = <54 4>; + interrupt-controller; + #interrupt-cells = <1>; + #clock-cells = <0>; + clock-frequency = <9600000>; + clock-output-names = "mclk"; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headset-vthreshold-microvolt = <1700000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + clock-names = "extclk"; + clocks = <&rpmhcc 2>; + + #address-cells = <1>; + #size-cells = <1>; + + gpio@42 { + compatible = "qcom,wcd9340-gpio"; + reg = <0x42 0x2>; + gpio-controller; + #gpio-cells = <2>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml new file mode 100644 index 000000000000..49a267b306f6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm SoundWire Slave devices on WCD9380/WCD9385 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. This bindings is for the + slave devices. + +properties: + compatible: + const: sdw20217010d00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between slave and master tx ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between slave and master rx ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml new file mode 100644 index 000000000000..51547190f709 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -0,0 +1,153 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WCD9380/WCD9385 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. + +properties: + compatible: + enum: + - qcom,wcd9380-codec + - qcom,wcd9385-codec + + reset-gpios: + description: GPIO spec for reset line to use + maxItems: 1 + + us-euro-gpios: + description: GPIO spec for swapping gnd and mic segments + maxItems: 1 + + vdd-buck-supply: + description: A reference to the 1.8V buck supply + + vdd-rxtx-supply: + description: A reference to the 1.8V rx supply + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-mic-bias-supply: + description: A reference to the 3.8V mic bias supply + + qcom,tx-device: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A reference to Soundwire tx device phandle + + qcom,rx-device: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A reference to Soundwire rx device phandle + + qcom,micbias1-microvolt: + description: micbias1 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias2-microvolt: + description: micbias2 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias3-microvolt: + description: micbias3 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias4-microvolt: + description: micbias4 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,hphl-jack-type-normally-closed: + description: Indicates that HPHL jack switch type is normally closed + type: boolean + + qcom,ground-jack-type-normally-closed: + description: Indicates that Headset Ground switch type is normally closed + type: boolean + + qcom,mbhc-headset-vthreshold-microvolt: + description: Voltage threshold value for headset detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-headphone-vthreshold-microvolt: + description: Voltage threshold value for headphone detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-buttons-vthreshold-microvolt: + description: + Array of 8 Voltage threshold values corresponding to headset + button0 - button7 + minItems: 8 + maxItems: 8 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reset-gpios + - qcom,tx-device + - qcom,rx-device + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + codec { + compatible = "qcom,wcd9380-codec"; + reset-gpios = <&tlmm 32 0>; + #sound-dai-cells = <1>; + qcom,tx-device = <&wcd938x_tx>; + qcom,rx-device = <&wcd938x_rx>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + }; + + /* ... */ + + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml new file mode 100644 index 000000000000..ea44d03e58ca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa881x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WSA8810/WSA8815 Class-D Smart Speaker Amplifier + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + WSA8810 is a class-D smart speaker amplifier and WSA8815 + is a high-output power class-D smart speaker amplifier. + Their primary operating mode uses a SoundWire digital audio + interface. This binding is for SoundWire interface. + +properties: + compatible: + const: sdw10217201000 + + reg: + maxItems: 1 + + powerdown-gpios: + description: GPIO spec for Powerdown/Shutdown line to use + maxItems: 1 + + '#thermal-sensor-cells': + const: 0 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - powerdown-gpios + - "#thermal-sensor-cells" + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + soundwire@c2d0000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x0c2d0000 0x2000>; + + speaker@0,1 { + compatible = "sdw10217201000"; + reg = <0 1>; + powerdown-gpios = <&wcdpinctrl 2 0>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + + speaker@0,2 { + compatible = "sdw10217201000"; + reg = <0 2>; + powerdown-gpios = <&wcdpinctrl 2 0>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml new file mode 100644 index 000000000000..1d73204451b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015p.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt1015p codec devicetree bindings + +maintainers: + - Tzung-Bi Shih <tzungbi@google.com> + +description: | + Rt1015p is a rt1015 variant which does not support I2C and + only supports S24, 48kHz, 64FS. + +properties: + compatible: + enum: + - realtek,rt1015p + - realtek,rt1019p + + sdb-gpios: + description: + GPIO used for shutdown control. + 0 means shut down; 1 means power on. + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + rt1015p: rt1015p { + compatible = "realtek,rt1015p"; + sdb-gpios = <&pio 175 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml new file mode 100644 index 000000000000..ca5b8987b749 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -0,0 +1,121 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5682s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5682s codec devicetree bindings + +maintainers: + - Derek Fang <derek.fang@realtek.com> + +description: | + Rt5682s(ALC5682I-VS) is a rt5682i variant which supports I2C only. + +properties: + compatible: + const: realtek,rt5682s + + reg: + maxItems: 1 + description: I2C address of the device. + + interrupts: + maxItems: 1 + description: The CODEC's interrupt output. + + realtek,dmic1-data-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 data is not used + - 1 # using GPIO2 pin as dmic1 data pin + - 2 # using GPIO5 pin as dmic1 data pin + description: | + Specify which GPIO pin be used as DMIC1 data pin. + + realtek,dmic1-clk-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 clk is not used + - 1 # using GPIO1 pin as dmic1 clock pin + - 2 # using GPIO3 pin as dmic1 clock pin + description: | + Specify which GPIO pin be used as DMIC1 clk pin. + + realtek,jd-src: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # No JD is used + - 1 # using JD1 as JD source + description: | + Specify which JD source be used. + + realtek,ldo1-en-gpios: + description: | + The GPIO that controls the CODEC's LDO1_EN pin. + + realtek,dmic-clk-rate-hz: + description: | + Set the clock rate (hz) for the requirement of the particular DMIC. + + realtek,dmic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular DMIC. + + realtek,amic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular platform or AMIC. + + realtek,dmic-clk-driving-high: + type: boolean + description: | + Set the high driving of the DMIC clock out. + + clocks: + items: + - description: phandle and clock specifier for codec MCLK. + + clock-names: + items: + - const: mclk + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: Name given for DAI word clock and bit clock outputs. + +additionalProperties: false + +required: + - compatible + - reg + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,rt5682s"; + reg = <0x1a>; + interrupts = <6 IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio 2 GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt deleted file mode 100644 index 0cf0f819b823..000000000000 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt +++ /dev/null @@ -1,31 +0,0 @@ -Renesas FSI - -Required properties: -- compatible : "renesas,fsi2-<soctype>", - "renesas,sh_fsi2" or "renesas,sh_fsi" as - fallback. - Examples with soctypes are: - - "renesas,fsi2-r8a7740" (R-Mobile A1) - - "renesas,fsi2-sh73a0" (SH-Mobile AG5) -- reg : Should contain the register physical address and length -- interrupts : Should contain FSI interrupt - -- fsia,spdif-connection : FSI is connected by S/PDIF -- fsia,stream-mode-support : FSI supports 16bit stream mode. -- fsia,use-internal-clock : FSI uses internal clock when master mode. - -- fsib,spdif-connection : same as fsia -- fsib,stream-mode-support : same as fsia -- fsib,use-internal-clock : same as fsia - -Example: - -sh_fsi2: sh_fsi2@ec230000 { - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupts = <0 146 0x4>; - - fsia,spdif-connection; - fsia,stream-mode-support; - fsia,use-internal-clock; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml new file mode 100644 index 000000000000..0dd3f7361399 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,fsi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas FIFO-buffered Serial Interface (FSI) + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for FSI2 SoC + - items: + - enum: + - renesas,fsi2-sh73a0 # SH-Mobile AG5 + - renesas,fsi2-r8a7740 # R-Mobile A1 + - enum: + - renesas,sh_fsi2 + # for Generic + - items: + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + power-domains: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +patternProperties: + "^fsi(a|b),spdif-connection$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI is connected by S/PDIF + + "^fsi(a|b),stream-mode-support$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI supports 16bit stream mode + + "^fsi(a|b),use-internal-clock$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI uses internal clock when master mode + +required: + - compatible + - reg + - interrupts + - clocks + - power-domains + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/r8a7740-clock.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + sh_fsi2: sound@fe1f0000 { + compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2"; + reg = <0xfe1f0000 0x400>; + interrupts = <GIC_SPI 9 0x4>; + clocks = <&mstp3_clks R8A7740_CLK_FSI>; + power-domains = <&pd_a4mp>; + + #sound-dai-cells = <1>; + fsia,spdif-connection; + fsia,stream-mode-support; + fsia,use-internal-clock; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 5c52182f7dcf..b731f16aea84 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,520 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "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) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "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 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- 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. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - 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 - it must be 1 if your system has audio_clkout0/1/2/3 -- 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 -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- 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) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - 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>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - 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>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -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"; - - 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>; - }; - - 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>; - }; - }; -}; - -&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 -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 000000000000..e17c0245f77a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,453 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7742 # RZ/G1H + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a774e1 # RZ/G2H + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77961 # 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 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 1 + maxItems: 5 + + reg-names: + minItems: 1 + maxItems: 5 + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + minItems: 1 + maxItems: 11 + + reset-names: + minItems: 1 + maxItems: 11 + + clocks: + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + minItems: 1 + maxItems: 31 + + clock-names: + description: List of necessary clock names. + minItems: 1 + maxItems: 31 + items: + oneOf: + - const: ssi-all + - pattern: '^ssi\.[0-9]$' + - pattern: '^src\.[0-9]$' + - pattern: '^mix\.[0-1]$' + - pattern: '^ctu\.[0-1]$' + - pattern: '^dvc\.[0-1]$' + - pattern: '^clk_(a|b|c|i)$' + + ports: + $ref: /schemas/graph.yaml#/properties/ports + patternProperties: + port(@[0-9a-f]+)?: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + rcar_sound,dvc: + description: DVC subnode. + type: object + patternProperties: + "^dvc-[0-1]$": + type: object + properties: + dmas: + maxItems: 1 + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + rcar_sound,mix: + description: MIX subnode. + type: object + patternProperties: + "^mix-[0-1]$": + type: object + # no properties + additionalProperties: false + + rcar_sound,ctu: + description: CTU subnode. + type: object + patternProperties: + "^ctu-[0-7]$": + type: object + # no properties + additionalProperties: false + + rcar_sound,src: + description: SRC subnode. + type: object + patternProperties: + "^src-[0-9]$": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + maxItems: 2 + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + rcar_sound,ssiu: + description: SSIU subnode. + type: object + patternProperties: + "^ssiu-[0-9]+$": + type: object + properties: + dmas: + maxItems: 2 + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + rcar_sound,ssi: + description: SSI subnode. + type: object + patternProperties: + "^ssi-[0-9]$": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + minItems: 2 + maxItems: 4 + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + rcar_sound,dai: + description: DAI subnode. + type: object + patternProperties: + "^dai([0-9]+)?$": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +allOf: + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen1 + then: + properties: + reg: + maxItems: 3 + reg-names: + maxItems: 3 + items: + enum: + - scu + - ssi + - adg + else: + properties: + reg: + maxItems: 5 + reg-names: + maxItems: 5 + items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0xec500000 0x1000>, /* SCU */ + <0xec5a0000 0x100>, /* ADG */ + <0xec540000 0x1000>, /* SSIU */ + <0xec541000 0x1280>, /* SSI */ + <0xec740000 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* MIX1, MIX0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CTU1, CTU0 */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>, /* DVC0, DVC1 */ + <&audio_clk_a>, <&audio_clk_b>, /* CLKA, CLKB */ + <&audio_clk_c>, <&audio_clk_i>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "mix.1", "mix.0", + "ctu.1", "ctu.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + 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"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5>, <&src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* assume audio-graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0>, <&src0>, <&dvc0>; + capture = <&ssi1>, <&src1>, <&dvc1>; + }; + }; + }; + + + /* assume audio-graph */ + codec { + port { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml new file mode 100644 index 000000000000..7e8d252f7bca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rz-ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2) + +maintainers: + - Biju Das <biju.das.jz@bp.renesas.com> + +properties: + compatible: + items: + - enum: + - renesas,r9a07g044-ssi # RZ/G2{L,LC} + - renesas,r9a07g054-ssi # RZ/V2L + - const: renesas,rz-ssi + + reg: + maxItems: 1 + + interrupts: + maxItems: 4 + + interrupt-names: + items: + - const: int_req + - const: dma_rx + - const: dma_tx + - const: dma_rt + + clocks: + maxItems: 4 + + clock-names: + items: + - const: ssi + - const: ssi_sfr + - const: audio_clk1 + - const: audio_clk2 + + power-domains: + maxItems: 1 + + resets: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + description: + The first cell represents a phandle to dmac + The second cell specifies the encoded MID/RID values of the SSI port + connected to the DMA client and the slave channel configuration + parameters. + bits[0:9] - Specifies MID/RID value of a SSI channel as below + MID/RID value of SSI rx0 = 0x256 + MID/RID value of SSI tx0 = 0x255 + MID/RID value of SSI rx1 = 0x25a + MID/RID value of SSI tx1 = 0x259 + MID/RID value of SSI rt2 = 0x25f + MID/RID value of SSI rx3 = 0x262 + MID/RID value of SSI tx3 = 0x261 + bit[10] - HIEN = 1, Detects a request in response to the rising edge + of the signal + bit[11] - LVL = 0, Detects based on the edge + bits[12:14] - AM = 2, Bus cycle mode + bit[15] - TM = 0, Single transfer mode + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: rt + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - interrupts + - interrupt-names + - clocks + - clock-names + - resets + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/r9a07g044-cpg.h> + + ssi0: ssi@10049c00 { + compatible = "renesas,r9a07g044-ssi", + "renesas,rz-ssi"; + reg = <0x10049c00 0x400>; + interrupts = <GIC_SPI 326 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 327 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 328 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 329 IRQ_TYPE_EDGE_RISING>; + interrupt-names = "int_req", "dma_rx", "dma_tx", "dma_rt"; + clocks = <&cpg CPG_MOD R9A07G044_SSI0_PCLK2>, + <&cpg CPG_MOD R9A07G044_SSI0_PCLK_SFR>, + <&audio_clk1>, + <&audio_clk2>; + clock-names = "ssi", "ssi_sfr", "audio_clk1", "audio_clk2"; + power-domains = <&cpg>; + resets = <&cpg R9A07G044_SSI0_RST_M2_REG>; + dmas = <&dmac 0x2655>, + <&dmac 0x2656>; + dma-names = "tx", "rx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml new file mode 100644 index 000000000000..5655ca568240 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/richtek,rt9120.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Richtek RT9120 Class-D audio amplifier + +maintainers: + - ChiYuan Huang <cy_huang@richtek.com> + +description: | + The RT9120 is a high efficiency, I2S-input, stereo audio power amplifier + delivering 2*20W into 8 Ohm BTL speaker loads. It supports the wide input + voltage range from 4.5V to 26.4V to meet the need on most common + applications like as TV, monitors. home entertainment, electronic music + equipment. + +properties: + compatible: + enum: + - richtek,rt9120 + + reg: + description: I2C device address + maxItems: 1 + + pwdnn-gpios: + description: GPIO used for power down, low active + maxItems: 1 + + dvdd-supply: + description: | + Supply for the default on DVDD power, voltage domain must be 3P3V or 1P8V + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - dvdd-supply + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt9120@1a { + compatible = "richtek,rt9120"; + reg = <0x1a>; + pwdnn-gpios = <&gpio26 2 0>; + dvdd-supply = <&vdd_io_reg>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml new file mode 100644 index 000000000000..6a7c004bef17 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,i2s-tdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S/TDM Controller + +description: + The Rockchip I2S/TDM Controller is a Time Division Multiplexed + audio interface found in various Rockchip SoCs, allowing up + to 8 channels of audio over a serial interface. + +maintainers: + - Nicolas Frattaroli <frattaroli.nicolas@gmail.com> + +properties: + compatible: + enum: + - rockchip,px30-i2s-tdm + - rockchip,rk1808-i2s-tdm + - rockchip,rk3308-i2s-tdm + - rockchip,rk3568-i2s-tdm + - rockchip,rv1126-i2s-tdm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + maxItems: 2 + items: + enum: + - rx + - tx + + clocks: + minItems: 3 + items: + - description: clock for TX + - description: clock for RX + - description: AHB clock driving the interface + - description: + Parent clock for mclk_tx (only required when using mclk-calibrate) + - description: + Parent clock for mclk_rx (only required when using mclk-calibrate) + - description: + Clock for sample rates that are an integer multiple of 8000 + (only required when using mclk-calibrate) + - description: + Clock for sample rates that are an integer multiple of 11025 + (only required when using mclk-calibrate) + + clock-names: + minItems: 3 + items: + - const: mclk_tx + - const: mclk_rx + - const: hclk + - const: mclk_tx_src + - const: mclk_rx_src + - const: mclk_root0 + - const: mclk_root1 + + resets: + minItems: 1 + maxItems: 2 + description: resets for the tx and rx directions + + reset-names: + minItems: 1 + maxItems: 2 + items: + enum: + - tx-m + - rx-m + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + rockchip,trcm-sync-tx-only: + type: boolean + description: Use TX BCLK/LRCK for both TX and RX. + + rockchip,trcm-sync-rx-only: + type: boolean + description: Use RX BCLK/LRCK for both TX and RX. + + "#sound-dai-cells": + const: 0 + + rockchip,i2s-rx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of I2S RX sdis to I2S data bus lines. + By default, they are mapped one-to-one. + rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0. + maxItems: 4 + items: + enum: [0, 1, 2, 3] + + rockchip,i2s-tx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of I2S TX sdos to I2S data bus lines. + By default, they are mapped one-to-one. + rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0. + maxItems: 4 + items: + enum: [0, 1, 2, 3] + + rockchip,io-multiplex: + description: + Specify that the GPIO lines on the I2S bus are multiplexed such that + the direction (input/output) needs to be dynamically adjusted. + type: boolean + + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + - resets + - reset-names + - rockchip,grf + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3568-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + i2s@fe410000 { + compatible = "rockchip,rk3568-i2s-tdm"; + reg = <0x0 0xfe410000 0x0 0x1000>; + interrupts = <GIC_SPI 53 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru MCLK_I2S1_8CH_TX>, <&cru MCLK_I2S1_8CH_RX>, + <&cru HCLK_I2S1_8CH>; + clock-names = "mclk_tx", "mclk_rx", "hclk"; + dmas = <&dmac1 3>, <&dmac1 2>; + dma-names = "rx", "tx"; + resets = <&cru SRST_M_I2S1_8CH_TX>, <&cru SRST_M_I2S1_8CH_RX>; + reset-names = "tx-m", "rx-m"; + rockchip,trcm-sync-tx-only; + rockchip,grf = <&grf>; + #sound-dai-cells = <0>; + pinctrl-names = "default"; + pinctrl-0 = + <&i2s1m0_sclktx + &i2s1m0_sclkrx + &i2s1m0_lrcktx + &i2s1m0_lrckrx + &i2s1m0_sdi0 + &i2s1m0_sdi1 + &i2s1m0_sdi2 + &i2s1m0_sdi3 + &i2s1m0_sdo0 + &i2s1m0_sdo1 + &i2s1m0_sdo2 + &i2s1m0_sdo3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.txt b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt deleted file mode 100644 index 98572a25122f..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,pdm.txt +++ /dev/null @@ -1,46 +0,0 @@ -* Rockchip PDM controller - -Required properties: - -- compatible: "rockchip,pdm" - - "rockchip,px30-pdm" - - "rockchip,rk1808-pdm" - - "rockchip,rk3308-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 -- resets: a list of phandle + reset-specifer paris, one for each entry in reset-names. -- reset-names: reset names, should include "pdm-m". -- 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_sdi0 - &pdmm0_sdi1 - &pdmm0_sdi2 - &pdmm0_sdi3>; - pinctrl-1 = <&pdmm0_clk_sleep - &pdmm0_sdi0_sleep - &pdmm0_sdi1_sleep - &pdmm0_sdi2_sleep - &pdmm0_sdi3_sleep>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml new file mode 100644 index 000000000000..22e1cf6c0592 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,pdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip PDM controller + +description: + 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. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + enum: + - rockchip,pdm + - rockchip,px30-pdm + - rockchip,rk1808-pdm + - rockchip,rk3308-pdm + - rockchip,rk3568-pdm + - rockchip,rv1126-pdm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for PDM controller + - description: clock for PDM BUS + + clock-names: + items: + - const: pdm_clk + - const: pdm_hclk + + dmas: + maxItems: 1 + + dma-names: + items: + - const: rx + + power-domains: + maxItems: 1 + + resets: + items: + - description: reset for PDM controller + + reset-names: + items: + - const: pdm-m + + rockchip,path-map: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of PDM SDIx to PDM PATHx. + By default, they are mapped one-to-one. + maxItems: 4 + uniqueItems: true + items: + enum: [ 0, 1, 2, 3 ] + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3328-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + + pdm@ff040000 { + compatible = "rockchip,pdm"; + reg = <0x0 0xff040000 0x0 0x1000>; + interrupts = <GIC_SPI 82 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_PDM>, <&cru HCLK_PDM>; + clock-names = "pdm_clk", "pdm_hclk"; + dmas = <&dmac 16>; + dma-names = "rx"; + #sound-dai-cells = <0>; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pdmm0_clk + &pdmm0_sdi0 + &pdmm0_sdi1 + &pdmm0_sdi2 + &pdmm0_sdi3>; + pinctrl-1 = <&pdmm0_clk_sleep + &pdmm0_sdi0_sleep + &pdmm0_sdi1_sleep + &pdmm0_sdi2_sleep + &pdmm0_sdi3_sleep>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 2469588c7ccb..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,23 +0,0 @@ -* 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/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..75b3b33b5f1f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt deleted file mode 100644 index 54aefab71f2c..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ /dev/null @@ -1,49 +0,0 @@ -* Rockchip I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-i2s": for rk3066 - - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 - - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 - - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "i2s_hclk": clock for I2S BUS - - "i2s_clk" : clock for I2S controller -- rockchip,playback-channels: max playback channels, if not set, 8 channels default. -- rockchip,capture-channels: max capture channels, if not set, 2 channels default. - -Required properties for controller which support multi channels -playback/capture: - -- rockchip,grf: the phandle of the syscon node for GRF register. - -Example for rk3288 I2S controller: - -i2s@ff890000 { - compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; - reg = <0xff890000 0x10000>; - interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&pdma1 0>, <&pdma1 1>; - dma-names = "tx", "rx"; - clock-names = "i2s_hclk", "i2s_clk"; - clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; - rockchip,playback-channels = <8>; - rockchip,capture-channels = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000000..5ea16b8ef93f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,125 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk1808-i2s + - rockchip,rk3036-i2s + - rockchip,rk3128-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3308-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - rockchip,rv1126-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + oneOf: + - const: rx + - items: + - const: tx + - const: rx + + power-domains: + maxItems: 1 + + reset-names: + items: + - const: reset-m + - const: reset-h + + resets: + maxItems: 2 + + rockchip,capture-channels: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3288-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98090.txt b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt index a805aa99ad75..e9c58b204399 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-max98090.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt @@ -5,15 +5,38 @@ Required properties: - rockchip,model: The user-visible name of this sound complex - rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's connected to the CODEC -- rockchip,audio-codec: The phandle of the MAX98090 audio codec -- rockchip,headset-codec: The phandle of Ext chip for jack detection + +Optional properties: +- rockchip,audio-codec: The phandle of the MAX98090 audio codec. +- rockchip,headset-codec: The phandle of Ext chip for jack detection. This is + required if there is rockchip,audio-codec. +- rockchip,hdmi-codec: The phandle of HDMI device for HDMI codec. Example: +/* For max98090-only board. */ +sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&max98090>; + rockchip,headset-codec = <&headsetcodec>; +}; + +/* For HDMI-only board. */ +sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,hdmi-codec = <&hdmi>; +}; + +/* For max98090 plus HDMI board. */ sound { compatible = "rockchip,rockchip-audio-max98090"; rockchip,model = "ROCKCHIP-I2S"; rockchip,i2s-controller = <&i2s>; rockchip,audio-codec = <&max98090>; rockchip,headset-codec = <&headsetcodec>; + rockchip,hdmi-codec = <&hdmi>; }; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271e92..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -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" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000000..d0a24bf928d6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - const: rockchip,rk3568-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - rockchip,rk3308-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + power-domains: + maxItems: 1 + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..859ce64da152 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro@katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt index 35a23e60d679..02d53b9aa247 100644 --- a/Documentation/devicetree/bindings/sound/rt1011.txt +++ b/Documentation/devicetree/bindings/sound/rt1011.txt @@ -20,6 +20,14 @@ Required properties: | 1 | 1 | 0x3b | ------------------------------------- +Optional properties: + +- realtek,temperature_calib + u32. The temperature was measured while doing the calibration. Units: Celsius degree + +- realtek,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + Pins on the device (for linking into audio routes) for RT1011: * SPO @@ -29,4 +37,6 @@ Example: rt1011: codec@38 { compatible = "realtek,rt1011"; reg = <0x38>; + realtek,temperature_calib = <25>; + realtek,r0_calib = <0x224050>; }; diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt new file mode 100644 index 000000000000..e498966d436f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -0,0 +1,23 @@ +RT1015 Mono Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1015". + +- reg : The I2C address of the device. + +Optional properties: + +- realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. + +Example: + +rt1015: codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt1016.txt b/Documentation/devicetree/bindings/sound/rt1016.txt new file mode 100644 index 000000000000..2310f8ff259b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1016.txt @@ -0,0 +1,17 @@ +RT1016 Stereo Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1016". + +- reg : The I2C address of the device. + + +Example: + +rt1016: codec@1a { + compatible = "realtek,rt1016"; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt1019.yaml b/Documentation/devicetree/bindings/sound/rt1019.yaml new file mode 100644 index 000000000000..3d5a91a942f4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1019.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rt1019.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1019 Mono Class-D Audio Amplifier + +maintainers: + - jack.yu@realtek.com + +properties: + compatible: + const: realtek,rt1019 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt1019: codec@28 { + compatible = "realtek,rt1019"; + reg = <0x28>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1308.txt b/Documentation/devicetree/bindings/sound/rt1308.txt index 2d46084afce4..2d46084afce4 100755..100644 --- a/Documentation/devicetree/bindings/sound/rt1308.txt +++ b/Documentation/devicetree/bindings/sound/rt1308.txt diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..ff1228713f7e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -88,7 +88,7 @@ rt5640 { compatible = "realtek,rt5640"; reg = <0x1c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5645.txt b/Documentation/devicetree/bindings/sound/rt5645.txt index a03f9a872a71..41a62fd2ae1f 100644 --- a/Documentation/devicetree/bindings/sound/rt5645.txt +++ b/Documentation/devicetree/bindings/sound/rt5645.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: - hp-detect-gpios: diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..013f534fa059 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -37,10 +37,21 @@ Optional properties: - realtek,jd-src 0: No JD is used 1: using JD3 as JD source + 2: JD source for Intel HDA header - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. - realtek,reset-gpios : The GPIO that controls the CODEC's RESET pin. +- sound-name-prefix: Please refer to name-prefix.yaml + +- ports: A Codec may have a single or multiple I2S interfaces. These + interfaces on Codec side can be described under 'ports' or 'port'. + When the SoC or host device is connected to multiple interfaces of + the Codec, the connectivity can be described using 'ports' property. + If a single interface is used, then 'port' can be used. The usage + depends on the platform or board design. + Please refer to Documentation/devicetree/bindings/graph.txt + Pins on the device (for linking into audio routes) for RT5659/RT5658: * DMIC L1 @@ -72,7 +83,7 @@ rt5659 { compatible = "realtek,rt5659"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 8df170506986..f6ca96b4ce98 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -62,7 +62,7 @@ rt5659 { compatible = "realtek,rt5665"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt index c88b96e7764b..a2b7e9a2f2f3 100644 --- a/Documentation/devicetree/bindings/sound/rt5668.txt +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -41,7 +41,7 @@ rt5668 { compatible = "realtek,rt5668b"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 1b3c13d206ff..da2430099181 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -64,7 +64,7 @@ rt5677 { compatible = "realtek,rt5677"; reg = <0x2c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; gpio-controller; #gpio-cells = <2>; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 312e9a129530..c5f2b8febcee 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -27,6 +27,27 @@ Optional properties: - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,btndet-delay + The debounce delay for push button. + The delay time is realtek,btndet-delay value multiple of 8.192 ms. + If absent, the default is 16. + +- #clock-cells : Should be set to '<1>', wclk and bclk sources provided. +- clock-output-names : Name given for DAI clocks output. + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- realtek,dmic-clk-rate-hz : Set the clock rate (hz) for the requirement of + the particular DMIC. + +- realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of + the particular DMIC. + +- realtek,dmic-clk-driving-high : Set the high driving of the DMIC clock out. + +- #sound-dai-cells: Should be set to '<0>'. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 @@ -41,10 +62,17 @@ rt5682 { compatible = "realtek,rt5682i"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; + realtek,btndet-delay = <16>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; }; diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml new file mode 100644 index 000000000000..a01c4ad929b8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -0,0 +1,151 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Aries audio complex with WM8994 codec + +maintainers: + - Jonathan Bakker <xc-racer2@live.ca> + +properties: + compatible: + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + minItems: 2 + maxItems: 2 + description: | + phandles to the I2S controller and bluetooth codec, + in that order + required: + - sound-dai + + codec: + type: object + properties: + sound-dai: + maxItems: 1 + description: phandle to the WM8994 CODEC + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and sinks are the WM8994's pins (as + documented in its binding), and the jacks on the board - + For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic, + or FM In + For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic, + or HeadsetMic + + extcon: + description: Extcon phandle for dock detection + + main-micbias-supply: + description: Supply for the micbias on the main mic + + headset-micbias-supply: + description: Supply for the micbias on the headset mic + + earpath-sel-gpios: + maxItems: 1 + description: GPIO for switching between tv-out and mic paths + + headset-detect-gpios: + maxItems: 1 + description: GPIO for detection of headset insertion + + headset-key-gpios: + maxItems: 1 + description: GPIO for detection of headset key press + + io-channels: + maxItems: 1 + description: IO channel to read micbias voltage for headset detection + + io-channel-names: + const: headset-detect + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - extcon + - main-micbias-supply + - headset-micbias-supply + - earpath-sel-gpios + - headset-detect-gpios + - headset-key-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,fascinate4g-wm8994"; + + model = "Fascinate4G"; + + extcon = <&fsa9480>; + + main-micbias-supply = <&main_micbias_reg>; + headset-micbias-supply = <&headset_micbias_reg>; + + earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>; + + io-channels = <&adc 3>; + io-channel-names = "headset-detect"; + headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>; + headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "LINE", "LINEOUT2N", + "LINE", "LINEOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + + "IN1RP", "Headset Mic", + "IN1RN", "Headset Mic"; + + pinctrl-names = "default"; + pinctrl-0 = <&headset_det &earpath_sel>; + + cpu { + sound-dai = <&i2s0>, <&bt_codec>; + }; + + codec { + sound-dai = <&wm8994>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,arndale.yaml b/Documentation/devicetree/bindings/sound/samsung,arndale.yaml new file mode 100644 index 000000000000..9bc4585bb6e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,arndale.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,arndale.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Insignal Arndale boards audio complex + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + enum: + - samsung,arndale-alc5631 + - samsung,arndale-rt5631 + - samsung,arndale-wm1811 + + samsung,audio-codec: + description: Phandle to the audio codec. + $ref: /schemas/types.yaml#/definitions/phandle + + samsung,audio-cpu: + description: Phandle to the Samsung I2S controller. + $ref: /schemas/types.yaml#/definitions/phandle + + samsung,model: + description: The user-visible name of this sound complex. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - samsung,audio-codec + - samsung,audio-cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,arndale-rt5631"; + samsung,audio-cpu = <&i2s0>; + samsung,audio-codec = <&rt5631>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 000000000000..ec50bcb4af5f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,110 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + maxItems: 1 + description: phandle to the I2S controller + required: + - sound-dai + + codec: + type: object + properties: + sound-dai: + maxItems: 1 + description: phandle to the WM1811 CODEC + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + maxItems: 1 + description: GPIO pin for FM selection + + lineout-sel-gpios: + maxItems: 1 + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt deleted file mode 100644 index e9da2200e173..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ /dev/null @@ -1,54 +0,0 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec - -Required properties: - - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), - "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), - "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - - model - the user-visible name of this sound complex - - clocks - should contain entries matching clock names in the clock-names - property - - samsung,audio-widgets - this property specifies off-codec audio elements - like headphones or speakers, for details see widgets.txt - - samsung,audio-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 and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - - For Odroid X2: - "Headphone Jack", "Mic Jack", "DMIC" - - For Odroid U3, XU3: - "Headphone Jack", "Speakers" - - 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 = "hardkernel,odroid-xu3-audio"; - model = "Odroid-XU3"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; - - cpu { - sound-dai = <&i2s0 0>; - }; - codec { - sound-dai = <&hdmi>, <&max98090>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000000..7b4e08ddef6a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + clocks: true + + cpu: + type: object + properties: + sound-dai: + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + items: + - description: phandle of the MAX98090 CODEC + - description: phandle of the HDMI IP block node + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + +required: + - compatible + - model + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt deleted file mode 100644 index 4686646fb122..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt +++ /dev/null @@ -1,14 +0,0 @@ -Samsung SMDK audio complex - -Required properties: -- compatible : "samsung,smdk-wm8994" -- samsung,i2s-controller: The phandle of the Samsung I2S0 controller -- samsung,audio-codec: The phandle of the WM8994 audio codec -Example: - -sound { - compatible = "samsung,smdk-wm8994"; - - samsung,i2s-controller = <&i2s0>; - samsung,audio-codec = <&wm8994>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml new file mode 100644 index 000000000000..ac151d3c1d77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,smdk5250.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SMDK5250 audio complex with WM8994 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + const: samsung,smdk-wm8994 + + samsung,audio-codec: + description: Phandle to the audio codec. + $ref: /schemas/types.yaml#/definitions/phandle + + samsung,i2s-controller: + description: Phandle to the Samsung I2S controller. + $ref: /schemas/types.yaml#/definitions/phandle + +required: + - compatible + - samsung,audio-codec + - samsung,i2s-controller + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,smdk-wm8994"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&wm8994>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml new file mode 100644 index 000000000000..51a83d3c7274 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,snow.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google Snow audio complex with MAX9809x codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + enum: + - google,snow-audio-max98090 + - google,snow-audio-max98091 + - google,snow-audio-max98095 + + codec: + type: object + properties: + sound-dai: + description: List of phandles to the CODEC and HDMI IP nodes. + items: + - description: Phandle to the MAX98090, MAX98091 or MAX98095 CODEC. + - description: Phandle to the HDMI IP block node. + required: + - sound-dai + + cpu: + type: object + properties: + sound-dai: + description: Phandle to the Samsung I2S controller. + maxItems: 1 + required: + - sound-dai + + samsung,audio-codec: + description: Phandle to the audio codec. + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + + samsung,i2s-controller: + description: Phandle to the Samsung I2S controller. + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + + samsung,model: + description: The user-visible name of this sound complex. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - codec + - cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&max98095 0>, <&hdmi>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-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 and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml new file mode 100644 index 000000000000..491e08019c04 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + audio-amplifier: + description: Phandle to the MAX98504 amplifier. + $ref: /schemas/types.yaml#/definitions/phandle + + audio-codec: + description: Phandles to the codecs. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - description: Phandle to the WM5110 audio codec. + - description: Phandle to the HDMI transmitter node. + + samsung,audio-routing: + description: | + 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 and sinks are the + WM5110's and MAX98504's pins and the jacks on the board: HP, SPK, Main + Mic, Sub Mic, Third Mic, Headset Mic. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + i2s-controller: + description: Phandles to the I2S controllers. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - description: Phandle to I2S0. + - description: Phandle to I2S1. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + + model: + description: The user-visible name of this sound complex. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - audio-amplifier + - audio-codec + - samsung,audio-routing + - i2s-controller + - mic-bias-gpios + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 GPIO_ACTIVE_HIGH>; + model = "wm5110"; + samsung,audio-routing = "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "RCV", "HPOUT3L", + "RCV", "HPOUT3R"; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: 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. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..84c4d6cba521 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,149 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with + secondary FIFO, s/w reset control and internal mux for root clock + source. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for + playback, stereo channel capture, secondary FIFO using internal + or external DMA, s/w reset control, internal mux for root clock + source and 7.1 channel TDM support for playback; TDM (Time division + multiplexing) is to allow transfer of multiple channel audio data on + single data line. + + samsung,exynos7-i2s: with all the available features of Exynos5 I2S. + Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO + with only external DMA and more number of root clock sampling + frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + assigned-clock-parents: true + assigned-clocks: true + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for I2S0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for I2S1 and I2S2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clock. I2S0 has internal mux to select the source + of root clock and I2S1 and I2S2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for I2S0 + - const: i2s_cdclk0 + - items: # for I2S1 + - const: i2s_cdclk1 + - items: # for I2S2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + interrupts: + maxItems: 1 + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/serial-midi.yaml b/Documentation/devicetree/bindings/sound/serial-midi.yaml new file mode 100644 index 000000000000..4afc29376efc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/serial-midi.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause + +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/serial-midi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Generic Serial MIDI Interface + +maintainers: + - Daniel Kaehn <kaehndan@gmail.com> + +description: + Generic MIDI interface using a serial device. This denotes that a serial device is + dedicated to MIDI communication, either to an external MIDI device through a DIN5 + or other connector, or to a known hardwired MIDI controller. This device must be a + child node of a serial node. + + Can only be set to use standard baud rates corresponding to supported rates of the + parent serial device. If the standard MIDI baud of 31.25 kBaud is needed + (as would be the case if interfacing with arbitrary external MIDI devices), + configure the clocks of the parent serial device so that a requested baud of 38.4 kBaud + resuts in the standard MIDI baud rate, and set the 'current-speed' property to 38400 (default) + +properties: + compatible: + const: serial-midi + + current-speed: + description: Baudrate to set the serial port to when this MIDI device is opened. + default: 38400 + +required: + - compatible + +additionalProperties: false + +examples: + - | + serial { + midi { + compatible = "serial-midi"; + }; + }; + - | + serial { + midi { + compatible = "serial-midi"; + current-speed = <115200>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt deleted file mode 100644 index 9d9ff5184939..000000000000 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ /dev/null @@ -1,60 +0,0 @@ -* Freescale SGTL5000 Stereo Codec - -Required properties: -- compatible : "fsl,sgtl5000". - -- reg : the I2C address of the device - -- #sound-dai-cells: must be equal to 0 - -- clocks : the clock provider of SYS_MCLK - -- VDDA-supply : the regulator provider of VDDA - -- VDDIO-supply: the regulator provider of VDDIO - -Optional properties: - -- VDDD-supply : the regulator provider of VDDD - -- micbias-resistor-k-ohms : the bias resistor to be used in kOhms - The resistor can take values of 2k, 4k or 8k. - If set to 0 it will be off. - If this node is not mentioned or if the value is unknown, then - micbias resistor is set to 4K. - -- micbias-voltage-m-volts : the bias voltage to be used in mVolts - The voltage can take values from 1.25V to 3V by 250mV steps - If this node is not mentioned or the value is unknown, then - the value is set to 1.25V. - -- lrclk-strength: the LRCLK 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 - -- 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 { - compatible = "fsl,sgtl5000"; - reg = <0x0a>; - #sound-dai-cells = <0>; - clocks = <&clks 150>; - micbias-resistor-k-ohms = <2>; - micbias-voltage-m-volts = <2250>; - VDDA-supply = <®_3p3v>; - VDDIO-supply = <®_3p3v>; -}; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml new file mode 100644 index 000000000000..e762c320b574 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale SGTL5000 Stereo Codec + +maintainers: + - Fabio Estevam <festevam@gmail.com> + +properties: + compatible: + const: fsl,sgtl5000 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + + clocks: + items: + - description: the clock provider of SYS_MCLK + + VDDA-supply: + description: the regulator provider of VDDA + + VDDIO-supply: + description: the regulator provider of VDDIO + + VDDD-supply: + description: the regulator provider of VDDD + + micbias-resistor-k-ohms: + description: The bias resistor to be used in kOhms. The resistor can take + values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not + mentioned or if the value is unknown, then micbias resistor is set to + 4k. + enum: [ 0, 2, 4, 8 ] + + micbias-voltage-m-volts: + description: The bias voltage to be used in mVolts. The voltage can take + values from 1.25V to 3V by 250mV steps. If this node is not mentioned + or the value is unknown, then the value is set to 1.25V. + enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] + + lrclk-strength: + description: | + The LRCLK 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 + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + + sclk-strength: + description: | + 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 + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <®_3p3v>; + VDDIO-supply = <®_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt deleted file mode 100644 index b1b097cc9b68..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ /dev/null @@ -1,17 +0,0 @@ -Simple Amplifier Audio Driver - -Required properties: -- compatible : "dioo,dio2125" or "simple-audio-amplifier" - -Optional properties: -- enable-gpios : the gpio connected to the enable pin of the simple amplifier -- VCC-supply : power supply for the device, as covered - in Documentation/devicetree/bindings/regulator/regulator.txt - -Example: - -amp: analog-amplifier { - compatible = "simple-audio-amplifier"; - VCC-supply = <®ulator>; - enable-gpios = <&gpio GPIOH_3 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml new file mode 100644 index 000000000000..8327846356d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-amplifier.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Amplifier Device Tree Bindings + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + enum: + - dioo,dio2125 + - simple-audio-amplifier + + enable-gpios: + maxItems: 1 + + VCC-supply: + description: > + power supply for the device + + sound-name-prefix: true + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/meson8-gpio.h> + + analog-amplifier { + compatible = "simple-audio-amplifier"; + VCC-supply = <®ulator>; + enable-gpios = <&gpio GPIOH_3 0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000000..b5fc35ee9b65 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-mux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Multiplexer + +maintainers: + - Alexandre Belloni <aleandre.belloni@bootlin.com> + +description: | + Simple audio multiplexers are driven using gpios, allowing to select which of + their input line is connected to the output line. + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: simple-audio-mux + + mux-gpios: + description: | + GPIOs used to select the input line. + + sound-name-prefix: true + +required: + - compatible + - mux-gpios + +additionalProperties: false + +examples: + - | + mux { + compatible = "simple-audio-mux"; + mux-gpios = <&gpio 3 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- 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. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- 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(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - 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-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..b261d49b9ddb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,500 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + + system-clock-fixed: + description: | + Specifies that the clock frequency should not be modified. + Implied when system-clock-frequency is specified, but can be used when + a clock is mapped to the device whose frequency cannot or should not be + changed. When mclk-fs is also specified, this restricts the device to a + single fixed sampling rate. + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + label: + maxItems: 1 + + routing: + description: | + 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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + $ref: "#/definitions/dai-tdm-slot-num" + dai-tdm-slot-width: + $ref: "#/definitions/dai-tdm-slot-width" + clocks: + maxItems: 1 + system-clock-frequency: + $ref: "#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "#/definitions/system-clock-direction-out" + system-clock-fixed: + $ref: "#/definitions/system-clock-fixed" + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": + const: 1 + "#size-cells": + const: 0 + + label: + $ref: "#/definitions/label" + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + $ref: "#/definitions/widgets" + simple-audio-card,routing: + $ref: "#/definitions/routing" + + # common properties + simple-audio-card,frame-master: + $ref: "#/definitions/frame-master" + simple-audio-card,bitclock-master: + $ref: "#/definitions/bitclock-master" + simple-audio-card,frame-inversion: + $ref: "#/definitions/frame-inversion" + simple-audio-card,bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + simple-audio-card,format: + $ref: "#/definitions/format" + simple-audio-card,mclk-fs: + $ref: "#/definitions/mclk-fs" + simple-audio-card,aux-devs: + $ref: "#/definitions/aux-devs" + simple-audio-card,convert-rate: + $ref: "#/definitions/convert-rate" + simple-audio-card,convert-channels: + $ref: "#/definitions/convert-channels" + simple-audio-card,prefix: + $ref: "#/definitions/prefix" + simple-audio-card,pin-switches: + $ref: "#/definitions/pin-switches" + simple-audio-card,hp-det-gpio: + maxItems: 1 + simple-audio-card,mic-det-gpio: + maxItems: 1 + +patternProperties: + "^simple-audio-card,cpu(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + maxItems: 1 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + + patternProperties: + "^cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# 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>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; + + 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,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: 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 { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi 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"; + #address-cells = <1>; + #size-cells = <0>; + + 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 { + reg = <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 { + reg = <1>; + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu0 { + sound-dai = <&rcar_sound 1>; + }; + cpu1 { + sound-dai = <&rcar_sound 2>; + }; + cpu2 { + sound-dai = <&rcar_sound 3>; + }; + cpu3 { + 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 { + reg = <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/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt deleted file mode 100644 index 062f5ec36f9b..000000000000 --- a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt +++ /dev/null @@ -1,17 +0,0 @@ -SiRF internal audio CODEC - -Required properties: - - - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec" - - - reg : the register address of the device. - - - clocks: the clock of SiRF internal audio codec - -Example: - -audiocodec: audiocodec@b0040000 { - compatible = "sirf,atlas6-audio-codec"; - reg = <0xb0040000 0x10000>; - clocks = <&clks 27>; -}; diff --git a/Documentation/devicetree/bindings/sound/sirf-usp.txt b/Documentation/devicetree/bindings/sound/sirf-usp.txt deleted file mode 100644 index 02f85b32d359..000000000000 --- a/Documentation/devicetree/bindings/sound/sirf-usp.txt +++ /dev/null @@ -1,27 +0,0 @@ -* SiRF SoC USP module - -Required properties: -- compatible: "sirf,prima2-usp-pcm" -- reg: Base address and size entries: -- dmas: 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. - - One of the DMA channels will be responsible for transmission (should be - named "tx") and one for reception (should be named "rx"). - -- clocks: USP controller clock source -- pinctrl-names: Must contain a "default" entry. -- pinctrl-NNN: One property must exist for each entry in pinctrl-names. - -Example: -usp0: usp@b0080000 { - compatible = "sirf,prima2-usp-pcm"; - reg = <0xb0080000 0x10000>; - clocks = <&clks 28>; - dmas = <&dmac1 1>, <&dmac1 2>; - dma-names = "rx", "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&usp0_only_utfs_pins_a>; -}; - diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt deleted file mode 100644 index 80fd9a87bb3f..000000000000 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ /dev/null @@ -1,31 +0,0 @@ -Audio Binding for Snow boards - -Required properties: -- compatible : Can be one of the following, - "google,snow-audio-max98090" or - "google,snow-audio-max98091" or - "google,snow-audio-max98095" -- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller -- samsung,audio-codec (deprecated): The phandle of the audio codec - -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 the phandle of the MAX98090, - MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible - string) and the second entry must be the phandle of the HDMI IP block node - -Optional: -- samsung,model: The name of the sound-card - -Example: - -sound { - compatible = "google,snow-audio-max98095"; - - samsung,model = "Snow-I2S-MAX98095"; - samsung,i2s-controller = <&i2s0>; - samsung,audio-codec = <&max98095>; -}; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000000..70f62ecd6eb2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel@alsa-project.org> + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: "/schemas/types.yaml#/definitions/phandle" + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels that + depends on SoC. + ld11-aio,ld20-aio: + 0: hdmi + 1: pcmin2 + 2: line + 3: hpcmout1 + 4: pcmout3 + 5: hiecout1 + 6: epcmout2 + 7: epcmout3 + 8: hieccompout1 + pxs2-aio: + 0: hdmi + 1: line + 2: aux + 3: hiecout1 + 4: iecout1 + 5: hieccompout1 + 6: ieccompout1 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000000..be6acfda9999 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel@alsa-project.org> + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels. + 0: line1 + 1: hp + 2: line2 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +additionalProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/sound-dai.yaml b/Documentation/devicetree/bindings/sound/sound-dai.yaml new file mode 100644 index 000000000000..61c6f7abc4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-dai.yaml @@ -0,0 +1,20 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface consumer Device Tree Bindings + +maintainers: + - Rob Herring <robh@kernel.org> + +select: true + +properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A phandle plus args to digital audio interface provider(s) + +additionalProperties: true +... diff --git a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt b/Documentation/devicetree/bindings/sound/spdif-transmitter.txt deleted file mode 100644 index 55a85841dd85..000000000000 --- a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt +++ /dev/null @@ -1,10 +0,0 @@ -Device-Tree bindings for dummy spdif transmitter - -Required properties: - - compatible: should be "linux,spdif-dit". - -Example node: - - codec: spdif-transmitter { - compatible = "linux,spdif-dit"; - }; diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index 4d51f3f5ea98..a6ffcdec6f6a 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -5,7 +5,7 @@ codec or external codecs. sti sound drivers allows to expose sti SoC audio interface through the generic ASoC simple card. For details about sound card declaration please refer to -Documentation/devicetree/bindings/sound/simple-card.txt. +Documentation/devicetree/bindings/sound/simple-card.yaml. 1) sti-uniperiph-dai: audio dai device. --------------------------------------- diff --git a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt deleted file mode 100644 index 864f5b00b031..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt +++ /dev/null @@ -1,63 +0,0 @@ -STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM) - -The DFSDM allows PDM microphones capture through SPI interface. The Audio -interface is seems as a sub block of the DFSDM device. -For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.txt - -Required properties: - - compatible: "st,stm32h7-dfsdm-dai". - - - #sound-dai-cells : Must be equal to 0 - - - io-channels : phandle to iio dfsdm instance node. - -Example of a sound card using audio DFSDM node. - - sound_card { - compatible = "audio-graph-card"; - - dais = <&cpu_port>; - }; - - dfsdm: dfsdm@40017000 { - compatible = "st,stm32h7-dfsdm"; - reg = <0x40017000 0x400>; - clocks = <&rcc DFSDM1_CK>; - clock-names = "dfsdm"; - #interrupt-cells = <1>; - #address-cells = <1>; - #size-cells = <0>; - - dfsdm_adc0: filter@0 { - compatible = "st,stm32-dfsdm-dmic"; - reg = <0>; - interrupts = <110>; - dmas = <&dmamux1 101 0x400 0x00>; - dma-names = "rx"; - st,adc-channels = <1>; - st,adc-channel-names = "dmic0"; - st,adc-channel-types = "SPI_R"; - st,adc-channel-clk-src = "CLKOUT"; - st,filter-order = <5>; - - dfsdm_dai0: dfsdm-dai { - compatible = "st,stm32h7-dfsdm-dai"; - #sound-dai-cells = <0>; - io-channels = <&dfsdm_adc0 0>; - cpu_port: port { - dfsdm_endpoint: endpoint { - remote-endpoint = <&dmic0_endpoint>; - }; - }; - }; - }; - - dmic0: dmic@0 { - compatible = "dmic-codec"; - #sound-dai-cells = <0>; - port { - dmic0_endpoint: endpoint { - remote-endpoint = <&dfsdm_endpoint>; - }; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt deleted file mode 100644 index cbf24bcd1b8d..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ /dev/null @@ -1,62 +0,0 @@ -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/st,stm32-pinctrl.yaml - -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>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000000..d3966ae04ad0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: i2sclk + - const: x8k + - const: x11k + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + "#clock-cells": + description: Configure the I2S device as MCLK clock provider. + const: 0 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -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" 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. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.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"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..fe2e15504ebc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,194 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + maxItems: 3 + + clock-names: + maxItems: 3 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: phandle of the SAI sub-block + - description: index of the SAI sub-block + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + $ref: /schemas/types.yaml#/definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + else: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai2: sai@4400b000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400b000 0x400>; + reg = <0x4400b000 0x4>, <0x4400b3f0 0x10>; + clocks = <&rcc SAI2>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "x8k", "x11k"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>; + pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>; + + sai2a: audio-controller@4400b004 { + #sound-dai-cells = <0>; + compatible = "st,stm32-sai-sub-a"; + reg = <0x4 0x1c>; + dmas = <&dmamux1 89 0x400 0x01>; + dma-names = "tx"; + clocks = <&rcc SAI2_K>; + clock-names = "sai_ck"; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt deleted file mode 100644 index 33826f2459fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt +++ /dev/null @@ -1,56 +0,0 @@ -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>; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 000000000000..837e830c47ac --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio data capture DMA + - description: IEC status bits capture DMA + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "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, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -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 - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -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. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt deleted file mode 100644 index 056a098495cc..000000000000 --- a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt +++ /dev/null @@ -1,14 +0,0 @@ -* Allwinner A64 Codec Analog Controls - -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/sun8i-a33-codec.txt b/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt deleted file mode 100644 index 2ca3d138528e..000000000000 --- a/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt +++ /dev/null @@ -1,63 +0,0 @@ -Allwinner SUN8I audio codec ------------------------------------- - -On Sun8i-A33 SoCs, the audio is separated in different parts: - - A DAI driver. It uses the "sun4i-i2s" driver which is - documented here: - Documentation/devicetree/bindings/sound/sun4i-i2s.txt - - An analog part of the codec which is handled as PRCM registers. - See Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt - - An digital part of the codec which is documented in this current - binding documentation. - - And finally, an audio card which links all the above components. - The simple-audio card will be used. - See Documentation/devicetree/bindings/sound/simple-card.txt - -This bindings documentation exposes Sun8i codec (digital part). - -Required properties: -- compatible: must be "allwinner,sun8i-a33-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain followings: - - "bus": the parent APB clock for this controller - - "mod": the parent module clock - -Here is an example to add a sound card and the codec binding on sun8i SoCs that -are similar to A33 using simple-card: - - sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "sun8i-a33-audio"; - simple-audio-card,format = "i2s"; - simple-audio-card,frame-master = <&link_codec>; - simple-audio-card,bitclock-master = <&link_codec>; - simple-audio-card,mclk-fs = <512>; - simple-audio-card,aux-devs = <&codec_analog>; - simple-audio-card,routing = - "Left DAC", "Digital Left DAC", - "Right DAC", "Digital Right DAC"; - - simple-audio-card,cpu { - sound-dai = <&dai>; - }; - - link_codec: simple-audio-card,codec { - sound-dai = <&codec>; - }; - - soc@1c00000 { - [...] - - audio-codec@1c22e00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun8i-a33-codec"; - reg = <0x01c22e00 0x400>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>; - clock-names = "bus", "mod"; - }; - }; - diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt deleted file mode 100644 index 07356758bd91..000000000000 --- a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt +++ /dev/null @@ -1,17 +0,0 @@ -* Allwinner Codec Analog Controls - -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 - -Example: -prcm: prcm@1f01400 { - codec_analog: codec-analog { - compatible = "allwinner,sun8i-a23-codec-analog"; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt index 2d71eb05c1d3..a7eecad83db1 100644 --- a/Documentation/devicetree/bindings/sound/tas2552.txt +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -33,4 +33,4 @@ tas2552: tas2552@41 { }; For more product information please see the link below: -http://www.ti.com/product/TAS2552 +https://www.ti.com/product/TAS2552 diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..5f7dd5d6cbca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,78 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy@ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2562 + https://www.ti.com/lit/gpn/tas2563 + https://www.ti.com/lit/gpn/tas2564 + https://www.ti.com/lit/gpn/tas2110 + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + - ti,tas2564 + - ti,tas2110 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..bc90e72bf7cf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704@thundersoft.com> + +description: | + The TAS2770 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x41 to 0x48. + + reset-gpio: + maxItems: 1 + description: GPIO used to reset the device. + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@41 { + compatible = "ti,tas2770"; + reg = <0x41>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 14 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml new file mode 100644 index 000000000000..66a0df8850ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020-2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas27xx.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2764/TAS2780 Smart PA + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated speaker voltage and current sense provides + for real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2764 + - ti,tas2780 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpios = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index 7481653fe8e3..df99ca9451b0 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -4,9 +4,9 @@ The TAS5720 serial control bus communicates through the I2C protocol only. The serial bus is also used for periodic codec fault checking/reporting during audio playback. For more product information please see the links below: -http://www.ti.com/product/TAS5720L -http://www.ti.com/product/TAS5720M -http://www.ti.com/product/TAS5722L +https://www.ti.com/product/TAS5720L +https://www.ti.com/product/TAS5720M +https://www.ti.com/product/TAS5722L Required properties: diff --git a/Documentation/devicetree/bindings/sound/tas5805m.yaml b/Documentation/devicetree/bindings/sound/tas5805m.yaml new file mode 100644 index 000000000000..3aade02d8a96 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas5805m.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tas5805m.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: TAS5805M audio amplifier + +maintainers: + - Daniel Beer <daniel.beer@igorinstitute.com> + +description: | + The TAS5805M is a class D audio amplifier with a built-in DSP. + +properties: + compatible: + enum: + - ti,tas5805m + + reg: + maxItems: 1 + description: | + I2C address of the amplifier. See the datasheet for possible values. + + pvdd-supply: + description: | + Regulator for audio power supply (PVDD in the datasheet). + + pdn-gpios: + description: | + Power-down control GPIO (PDN pin in the datasheet). + + ti,dsp-config-name: + description: | + The name of the DSP configuration that should be loaded for this + instance. Configuration blobs are sequences of register writes + generated from TI's PPC3 tool. + $ref: /schemas/types.yaml#/definitions/string + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + tas5805m: tas5805m@2c { + reg = <0x2c>; + compatible = "ti,tas5805m"; + + pvdd-supply = <&audiopwr>; + pdn-gpios = <&tlmm 160 0>; + + ti,dsp-config-name = "mono_pbtl_48khz"; + }; + }; + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 34cf70e2cbc4..4bb513ae62fc 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -14,8 +14,8 @@ For instance: dai-tdm-slot-tx-mask = <0 1>; dai-tdm-slot-rx-mask = <1 0>; -And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() -to specify a explicit mapping of the channels and the slots. If it's absent +And for each specified driver, there could be one .of_xlate_tdm_slot_mask() +to specify an explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. diff --git a/Documentation/devicetree/bindings/sound/test-component.yaml b/Documentation/devicetree/bindings/sound/test-component.yaml new file mode 100644 index 000000000000..17fdb4317239 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/test-component.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/test-component.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Test Component Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - test-cpu + - test-cpu-verbose + - test-cpu-verbose-dai + - test-cpu-verbose-component + - test-codec + - test-codec-verbose + - test-codec-verbose-dai + - test-codec-verbose-component + +required: + - compatible + +additionalProperties: true + +examples: + - | + test_cpu { + compatible = "test-cpu"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml new file mode 100644 index 000000000000..20ea5883b7ff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml @@ -0,0 +1,139 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi <peter.ujfalusi@ti.com> +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments J721e Common Processor Board Audio Support + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: | + The audio support on the board is using pcm3168a codec connected to McASP10 + serializers in parallel setup. + The pcm3168a SCKI clock is sourced from j721e AUDIO_REFCLK2 pin. + In order to support 48KHz and 44.1KHz family of sampling rates the parent + clock for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and + PLL15 (for 44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via + different HSDIVIDER. + + Clocking setup for j721e: + 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + + 44.1KHz family: + PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + + Clocking setup for j7200: + 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP0_AUXCLK ---> McASP0.auxclk + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + +properties: + compatible: + enum: + - ti,j721e-cpb-audio + - ti,j7200-cpb-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + ti,cpb-mcasp: + description: phandle to McASP used on CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,cpb-codec: + description: phandle to the pcm3168a codec used on the CPB + $ref: /schemas/types.yaml#/definitions/phandle + + clocks: + minItems: 4 + maxItems: 6 + + clock-names: + minItems: 4 + maxItems: 6 + +required: + - compatible + - model + - ti,cpb-mcasp + - ti,cpb-codec + - clocks + - clock-names + +additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: ti,j721e-cpb-audio + + then: + properties: + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: Parent for CPB_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + - description: Parent for CPB_SCKI clock (for 44.1KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-mcasp-auxclk-44100 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + - const: cpb-codec-scki-44100 + + - if: + properties: + compatible: + contains: + const: ti,j7200-cpb-audio + + then: + properties: + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + +examples: + - |+ + sound { + compatible = "ti,j721e-cpb-audio"; + model = "j721e-cpb"; + + ti,cpb-mcasp = <&mcasp10>; + ti,cpb-codec = <&pcm3168a_1>; + + clocks = <&k3_clks 184 1>, + <&k3_clks 184 2>, <&k3_clks 184 4>, + <&k3_clks 157 371>, + <&k3_clks 157 400>, <&k3_clks 157 401>; + clock-names = "cpb-mcasp-auxclk", + "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100", + "cpb-codec-scki", + "cpb-codec-scki-48000", "cpb-codec-scki-44100"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml new file mode 100644 index 000000000000..859d369c71e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi <peter.ujfalusi@ti.com> +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments J721e Common Processor Board Audio Support + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: | + The Infotainment board plugs into the Common Processor Board, the support of the + extension board is extending the CPB audio support, decribed in: + sound/ti,j721e-cpb-audio.txt + + The audio support on the Infotainment Expansion Board consists of McASP0 + connected to two pcm3168a codecs with dedicated set of serializers to each. + The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin. + + In order to support 48KHz and 44.1KHz family of sampling rates the parent clock + for AUDIO_REFCLK0 needs to be changed between PLL4 (for 48KHz) and PLL15 (for + 44.1KHz). The same PLLs are used for McASP0's AUXCLK clock via different + HSDIVIDER. + + Note: the same PLL4 and PLL15 is used by the audio support on the CPB! + + Clocking setup for 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + | |-> MCASP0_AUXCLK ---> McASP0.auxclk + | + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI + + Clocking setup for 44.1KHz family: + PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + | |-> MCASP0_AUXCLK ---> McASP0.auxclk + | + |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI + +properties: + compatible: + items: + - const: ti,j721e-cpb-ivi-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + ti,cpb-mcasp: + description: phandle to McASP used on CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,cpb-codec: + description: phandle to the pcm3168a codec used on the CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-mcasp: + description: phandle to McASP used on IVI + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-codec-a: + description: phandle to the pcm3168a-A codec on the expansion board + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-codec-b: + description: phandle to the pcm3168a-B codec on the expansion board + $ref: /schemas/types.yaml#/definitions/phandle + + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: Parent for CPB_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + - description: Parent for CPB_SCKI clock (for 44.1KHz) + - description: AUXCLK clock for McASP used by IVI audio + - description: Parent for IVI_McASP auxclk (for 48KHz) + - description: Parent for IVI_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on IVI + - description: Parent for IVI_SCKI clock (for 48KHz) + - description: Parent for IVI_SCKI clock (for 44.1KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-mcasp-auxclk-44100 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + - const: cpb-codec-scki-44100 + - const: ivi-mcasp-auxclk + - const: ivi-mcasp-auxclk-48000 + - const: ivi-mcasp-auxclk-44100 + - const: ivi-codec-scki + - const: ivi-codec-scki-48000 + - const: ivi-codec-scki-44100 + +required: + - compatible + - model + - ti,cpb-mcasp + - ti,cpb-codec + - ti,ivi-mcasp + - ti,ivi-codec-a + - ti,ivi-codec-b + - clocks + - clock-names + +additionalProperties: false + +examples: + - |+ + sound { + compatible = "ti,j721e-cpb-ivi-audio"; + model = "j721e-cpb-ivi"; + + ti,cpb-mcasp = <&mcasp10>; + ti,cpb-codec = <&pcm3168a_1>; + + ti,ivi-mcasp = <&mcasp0>; + ti,ivi-codec-a = <&pcm3168a_a>; + ti,ivi-codec-b = <&pcm3168a_b>; + + clocks = <&k3_clks 184 1>, + <&k3_clks 184 2>, <&k3_clks 184 4>, + <&k3_clks 157 371>, + <&k3_clks 157 400>, <&k3_clks 157 401>, + <&k3_clks 174 1>, + <&k3_clks 174 2>, <&k3_clks 174 4>, + <&k3_clks 157 301>, + <&k3_clks 157 330>, <&k3_clks 157 331>; + clock-names = "cpb-mcasp-auxclk", + "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100", + "cpb-codec-scki", + "cpb-codec-scki-48000", "cpb-codec-scki-44100", + "ivi-mcasp-auxclk", + "ivi-mcasp-auxclk-48000", "ivi-mcasp-auxclk-44100", + "ivi-codec-scki", + "ivi-codec-scki-48000", "ivi-codec-scki-44100"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt index 5d9cb84c661d..a02ecaab5183 100644 --- a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt +++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt @@ -25,6 +25,13 @@ Required properties: For required properties on SPI/I2C, consult SPI/I2C device tree documentation +Optional properties: + + - reset-gpios : Optional reset gpio line connected to RST pin of the codec. + The RST line is low active: + RST = low: device power-down + RST = high: device is enabled + Examples: i2c0: i2c0@0 { @@ -34,6 +41,7 @@ i2c0: i2c0@0 { pcm3168a: audio-codec@44 { compatible = "ti,pcm3168a"; reg = <0x44>; + reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>; clocks = <&clk_core CLK_AUDIO>; clock-names = "scki"; VDD1-supply = <&supply3v3>; diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt index eacb54f34188..00940c489299 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas6424.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt @@ -19,4 +19,4 @@ tas6424: tas6424@6a { }; For more product information please see the link below: -http://www.ti.com/product/TAS6424-Q1 +https://www.ti.com/product/TAS6424-Q1 diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..83936f594d1a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw@axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: audio-codec@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml new file mode 100644 index 000000000000..2ad17b361db0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -0,0 +1,209 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter + +maintainers: + - Dan Murphy <dmurphy@ti.com> + +description: | + The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital + PDM microphones recording), high-performance audio, analog-to-digital + converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140 + family supports line and microphone Inputs, and offers a programmable + microphone bias or supply voltage generation. + + Specifications can be found at: + https://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + +properties: + compatible: + enum: + - ti,tlv320adc3140 + - ti,tlv320adc5140 + - ti,tlv320adc6140 + + reg: + maxItems: 1 + description: | + I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpios: + maxItems: 1 + description: | + GPIO used for hardware reset. + + areg-supply: + description: | + Regulator with AVDD at 3.3V. If not defined then the internal regulator + is enabled. + + ti,mic-bias-source: + description: | + Indicates the source for MIC Bias. + 0 - Mic bias is set to VREF + 1 - Mic bias is set to VREF × 1.096 + 6 - Mic bias is set to AVDD + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 6] + + ti,vref-source: + description: | + Indicates the source for MIC Bias. + 0 - Set VREF to 2.75V + 1 - Set VREF to 2.5V + 2 - Set VREF to 1.375V + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + + ti,pdm-edge-select: + description: | + Defines the PDMCLK sampling edge configuration for the PDM inputs. This + array is defined as <PDMIN1 PDMIN2 PDMIN3 PDMIN4>. + + 0 - (default) Odd channel is latched on the negative edge and even + channel is latched on the the positive edge. + 1 - Odd channel is latched on the positive edge and even channel is + latched on the the negative edge. + + PDMIN1 - PDMCLK latching edge used for channel 1 and 2 data + PDMIN2 - PDMCLK latching edge used for channel 3 and 4 data + PDMIN3 - PDMCLK latching edge used for channel 5 and 6 data + PDMIN4 - PDMCLK latching edge used for channel 7 and 8 data + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + maximum: 1 + default: [0, 0, 0, 0] + + ti,gpi-config: + description: | + Defines the configuration for the general purpose input pins (GPI). + The array is defined as <GPI1 GPI2 GPI3 GPI4>. + + 0 - (default) disabled + 1 - GPIX is configured as a general-purpose input (GPI) + 2 - GPIX is configured as a master clock input (MCLK) + 3 - GPIX is configured as an ASI input for daisy-chain (SDIN) + 4 - GPIX is configured as a PDM data input for channel 1 and channel + (PDMDIN1) + 5 - GPIX is configured as a PDM data input for channel 3 and channel + (PDMDIN2) + 6 - GPIX is configured as a PDM data input for channel 5 and channel + (PDMDIN3) + 7 - GPIX is configured as a PDM data input for channel 7 and channel + (PDMDIN4) + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + maximum: 7 + default: [0, 0, 0, 0] + + ti,asi-tx-drive: + type: boolean + description: | + When set the device will set the Tx ASI output to a Hi-Z state for unused + data cycles. Default is to drive the output low on unused ASI cycles. + +patternProperties: + '^ti,gpo-config-[1-4]$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + Defines the configuration and output driver for the general purpose + output pins (GPO). These values are pairs, the first value is for the + configuration type and the second value is for the output drive type. + The array is defined as <GPO_CFG GPO_DRV> + + GPO output configuration can be one of the following: + + 0 - (default) disabled + 1 - GPOX is configured as a general-purpose output (GPO) + 2 - GPOX is configured as a device interrupt output (IRQ) + 3 - GPOX is configured as a secondary ASI output (SDOUT2) + 4 - GPOX is configured as a PDM clock output (PDMCLK) + + GPO output drive configuration for the GPO pins can be one of the following: + + 0d - (default) Hi-Z output + 1d - Drive active low and active high + 2d - Drive active low and weak high + 3d - Drive active low and Hi-Z + 4d - Drive weak low and active high + 5d - Drive Hi-Z and active high + + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tlv320adc5140"; + reg = <0x4c>; + ti,mic-bias-source = <6>; + ti,pdm-edge-select = <0 1 0 1>; + ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; + ti,gpo-config-1 = <0 0>; + ti,gpo-config-2 = <0 0>; + reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt index 5b3c33bb99e5..bbad98d5b986 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -29,6 +29,11 @@ Optional properties: 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD If this node is not mentioned or if the value is unknown, then micbias is set to 2.0V. +- ai31xx-ocmv - output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V Deprecated properties: @@ -53,7 +58,7 @@ The pins can be used in referring sound node's audio-routing property. Example: #include <dt-bindings/gpio/gpio.h> -#include <dt-bindings/sound/tlv320aic31xx-micbias.h> +#include <dt-bindings/sound/tlv320aic31xx.h> tlv320aic31xx: tlv320aic31xx@18 { compatible = "ti,tlv320aic311x"; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index ca75890f0d07..f59125bc79d1 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -6,6 +6,7 @@ Required properties: - compatible - "string" - One of: "ti,tlv320aic32x4" TLV320AIC3204 "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 + "ti,tas2505" TAS2505, TAS2521 - reg: I2C slave address - supply-*: Required supply regulators are: "iov" - digital IO power supply diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 9796c4639262..20931a63fd64 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -1,6 +1,6 @@ Texas Instruments - tlv320aic3x Codec module -The tlv320aic3x serial control bus communicates through I2C protocols +The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols Required properties: @@ -63,7 +63,7 @@ CODEC input pins for other compatible codecs: The pins can be used in referring sound node's audio-routing property. -Example: +I2C example: #include <dt-bindings/gpio/gpio.h> @@ -78,3 +78,20 @@ tlv320aic3x: tlv320aic3x@1b { DRVDD-supply = <®ulator>; DVDD-supply = <®ulator>; }; + +SPI example: + +spi0: spi@f0000000 { + tlv320aic3x: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + spi-max-frequency = <1000000>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + ai3x-ocmv = <0>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/uda1334.txt b/Documentation/devicetree/bindings/sound/uda1334.txt new file mode 100644 index 000000000000..f64071b25e8d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/uda1334.txt @@ -0,0 +1,17 @@ +UDA1334 audio CODEC + +This device uses simple GPIO pins for controlling codec settings. + +Required properties: + + - compatible : "nxp,uda1334" + - nxp,mute-gpios: a GPIO spec for the MUTE pin. + - nxp,deemph-gpios: a GPIO spec for the De-emphasis pin + +Example: + +uda1334: audio-codec { + compatible = "nxp,uda1334"; + nxp,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + nxp,deemph-gpios = <&gpio3 3 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt deleted file mode 100644 index 4ce68ed6f2f2..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,aio.txt +++ /dev/null @@ -1,45 +0,0 @@ -Socionext UniPhier SoC audio driver - -The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in -the same register space. - -Required properties: -- compatible : should be one of the following: - "socionext,uniphier-ld11-aio" - "socionext,uniphier-ld20-aio" - "socionext,uniphier-pxs2-aio" -- reg : offset and length of the register set for the device. -- interrupts : should contain I2S or S/PDIF interrupt. -- pinctrl-names : should be "default". -- pinctrl-0 : defined I2S signal pins for an external codec chip. -- clock-names : should include following entries: - "aio" -- clocks : a list of phandle, should contain an entry for each - entry in clock-names. -- reset-names : should include following entries: - "aio" -- resets : a list of phandle, should contain an entry for each - entry in reset-names. -- #sound-dai-cells: should be 1. - -Optional properties: -- socionext,syscon: a phandle, should contain soc-glue. - The soc-glue is used for changing mode of S/PDIF signal pin - to Output from Hi-Z. This property is optional if you use - I2S signal pins only. - -Example: - audio { - compatible = "socionext,uniphier-ld20-aio"; - reg = <0x56000000 0x80000>; - interrupts = <0 144 4>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_aout>; - clock-names = "aio"; - clocks = <&sys_clk 40>; - reset-names = "aio"; - resets = <&sys_rst 40>; - #sound-dai-cells = <1>; - - socionext,syscon = <&sg>; - }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt deleted file mode 100644 index 3f31b235f18b..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,evea.txt +++ /dev/null @@ -1,26 +0,0 @@ -Socionext EVEA - UniPhier SoC internal codec driver - -Required properties: -- compatible : should be "socionext,uniphier-evea". -- reg : offset and length of the register set for the device. -- clock-names : should include following entries: - "evea", "exiv" -- clocks : a list of phandle, should contain an entry for each - entries in clock-names. -- reset-names : should include following entries: - "evea", "exiv", "adamv" -- resets : a list of phandle, should contain reset entries of - reset-names. -- #sound-dai-cells: should be 1. - -Example: - - codec { - compatible = "socionext,uniphier-evea"; - reg = <0x57900000 0x1000>; - clock-names = "evea", "exiv"; - clocks = <&sys_clk 41>, <&sys_clk 42>; - reset-names = "evea", "exiv", "adamv"; - resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; - #sound-dai-cells = <1>; - }; diff --git a/Documentation/devicetree/bindings/sound/wlf,arizona.txt b/Documentation/devicetree/bindings/sound/wlf,arizona.txt deleted file mode 100644 index e172c62dc2df..000000000000 --- a/Documentation/devicetree/bindings/sound/wlf,arizona.txt +++ /dev/null @@ -1,53 +0,0 @@ -Cirrus Logic Arizona class audio SoCs - -These devices are audio SoCs with extensive digital capabilities and a range -of analogue I/O. - -This document lists sound specific bindings, see the primary binding -document: - ../mfd/arizona.txt - -Optional properties: - - - wlf,inmode : A list of INn_MODE register values, where n is the number - of input signals. Valid values are 0 (Differential), 1 (Single-ended) and - 2 (Digital Microphone). If absent, INn_MODE registers set to 0 by default. - If present, values must be specified less than or equal to the number of - input signals. If values less than the number of input signals, elements - that have not been specified are set to 0 by default. Entries are: - <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, wm8997) - <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814) - - wlf,out-mono : A list of boolean values indicating whether each output is - mono or stereo. Position within the list indicates the output affected - (eg. First entry in the list corresponds to output 1). A non-zero value - indicates a mono output. If present, the number of values should be less - than or equal to the number of outputs, if less values are supplied the - additional outputs will be treated as stereo. - - - wlf,dmic-ref : DMIC reference voltage source for each input, can be - selected from either MICVDD or one of the MICBIAS's, defines - (ARIZONA_DMIC_xxxx) are provided in <dt-bindings/mfd/arizona.txt>. If - present, the number of values should be less than or equal to the - number of inputs, unspecified inputs will use the chip default. - - - wlf,max-channels-clocked : The maximum number of channels to be clocked on - each AIF, useful for I2S systems with multiple data lines being mastered. - Specify one cell for each AIF to be configured, specify zero for AIFs that - should be handled normally. - If present, number of cells must be less than or equal to the number of - AIFs. If less than the number of AIFs, for cells that have not been - specified the corresponding AIFs will be treated as default setting. - - - wlf,spk-fmt : PDM speaker data format, must contain 2 cells (OUT5 and OUT6). - See the datasheet for values. - The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997, - wm8998, wm1814) - - - wlf,spk-mute : PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6). - See the datasheet for values. - The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997, - wm8998, wm1814) - - - wlf,out-volume-limit : The volume limit value that should be applied to each - output channel. See the datasheet for exact values. Channels are specified - in the order OUT1L, OUT1R, OUT2L, OUT2R, etc. diff --git a/Documentation/devicetree/bindings/sound/wlf,arizona.yaml b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml new file mode 100644 index 000000000000..1627c0bb69be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,arizona.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic/Wolfson Microelectronics Arizona class audio SoCs + +maintainers: + - patches@opensource.cirrus.com + +description: | + These devices are audio SoCs with extensive digital capabilities and a range + of analogue I/O. + + This document lists sound specific bindings, see the primary binding + document ../mfd/arizona.yaml + +properties: + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + wlf,inmode: + description: + A list of INn_MODE register values, where n is the number of input + signals. Valid values are 0 (Differential), 1 (Single-ended) and + 2 (Digital Microphone). If absent, INn_MODE registers set to 0 by + default. If present, values must be specified less than or equal + to the number of input signals. If values less than the number of + input signals, elements that have not been specified are set to 0 by + default. Entries are <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, + wm8997) and <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814) + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 2 + default: 0 + + wlf,out-mono: + description: + A list of boolean values indicating whether each output is mono + or stereo. Position within the list indicates the output affected + (eg. First entry in the list corresponds to output 1). A non-zero + value indicates a mono output. If present, the number of values + should be less than or equal to the number of outputs, if less values + are supplied the additional outputs will be treated as stereo. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 6 + items: + minimum: 0 + maximum: 1 + default: 0 + + wlf,dmic-ref: + description: + DMIC reference voltage source for each input, can be selected from + either MICVDD or one of the MICBIAS's, defines (ARIZONA_DMIC_xxxx) + are provided in dt-bindings/mfd/arizona.h. If present, the number + of values should be less than or equal to the number of inputs, + unspecified inputs will use the chip default. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 3 + default: 0 + + wlf,max-channels-clocked: + description: + The maximum number of channels to be clocked on each AIF, useful for + I2S systems with multiple data lines being mastered. Specify one + cell for each AIF to be configured, specify zero for AIFs that should + be handled normally. If present, number of cells must be less than + or equal to the number of AIFs. If less than the number of AIFs, for + cells that have not been specified the corresponding AIFs will be + treated as default setting. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 3 + items: + default: 0 + + wlf,spk-fmt: + description: + PDM speaker data format, must contain 2 cells (OUT5 and OUT6). See + the datasheet for values. The second cell is ignored for codecs that + do not have OUT6 (wm5102, wm8997, wm8998, wm1814) + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + wlf,spk-mute: + description: + PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6). See + the datasheet for values. The second cell is ignored for codecs that + do not have OUT6 (wm5102, wm8997, wm8998, wm1814) + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + wlf,out-volume-limit: + description: + The volume limit value that should be applied to each output + channel. See the datasheet for exact values. Channels are specified + in the order OUT1L, OUT1R, OUT2L, OUT2R, etc. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 12 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml new file mode 100644 index 000000000000..e7220e8b49f0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml @@ -0,0 +1,97 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson Microelectromics WM8731 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: | + Wolfson Microelectronics WM8731 audio CODEC + + Pins on the device (for linking into audio routes): + * LOUT: Left Channel Line Output + * ROUT: Right Channel Line Output + * LHPOUT: Left Channel Headphone Output + * RHPOUT: Right Channel Headphone Output + * LLINEIN: Left Channel Line Input + * RLINEIN: Right Channel Line Input + * MICIN: Microphone Input + +properties: + compatible: + enum: + - wlf,wm8731 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + description: Clock provider for MCLK pin. + maxItems: 1 + + clock-names: + items: + - const: mclk + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + HPVDD-supply: + description: Headphone power supply regulator on the HPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + spi-max-frequency: true + +additionalProperties: false + +required: + - reg + - compatible + - AVDD-supply + - HPVDD-supply + - DBVDD-supply + - DCVDD-supply + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + wm8731_i2c: codec@0 { + compatible = "wlf,wm8731"; + reg = <0>; + spi-max-frequency = <12500000>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; + - | + + i2c { + #address-cells = <1>; + #size-cells = <0>; + wm8731_spi: codec@1b { + compatible = "wlf,wm8731"; + reg = <0x1b>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000000..7105ed5fd6c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/wlf,wm8903.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: WM8903 audio codec + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8903 + + reg: + maxItems: 1 + + gpio-controller: true + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + micdet-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + description: Default register value for R6 (Mic Bias). + + micdet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 100 + description: The debounce delay for microphone detection in mS. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml new file mode 100644 index 000000000000..8aadcbeed502 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8940.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8940 Codec Device Tree Bindings + +maintainers: + - patches@opensource.cirrus.com + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8940 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml new file mode 100644 index 000000000000..5e172e9462b9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml @@ -0,0 +1,121 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8962.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8962 Ultra-Low Power Stereo CODEC + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8962 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + interrupts: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + AVDD-supply: + description: Analogue supply. + + CPVDD-supply: + description: Charge pump power supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + MICVDD-supply: + description: Microphone bias amp supply. + + PLLVDD-supply: + description: PLL Supply + + SPKVDD1-supply: + description: Supply for left speaker drivers. + + SPKVDD2-supply: + description: Supply for right speaker drivers. + + spk-mono: + $ref: /schemas/types.yaml#/definitions/flag + description: + If present, the SPK_MONO bit of R51 (Class D Control 2) gets set, + indicating that the speaker is in mono mode. + + mic-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Default register value for R48 (Additional Control 4). + If absent, the default should be the register default. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + description: + A list of GPIO configuration register values. If absent, no + configuration of these registers is performed. Note that only values + within [0x0, 0xffff] are valid. Any other value is regarded as setting + the GPIO register to its reset value 0x0. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + - PLLVDD-supply + - SPKVDD1-supply + - SPKVDD2-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx6qdl-clock.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8962: codec@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; + clocks = <&clks IMX6QDL_CLK_CKO>; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + CPVDD-supply = <®_audio>; + MICVDD-supply = <®_audio>; + PLLVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; + gpio-cfg = < + 0x0000 /* 0:Default */ + 0x0000 /* 1:Default */ + 0x0013 /* 2:FN_DMICCLK */ + 0x0000 /* 3:Default */ + 0x8014 /* 4:FN_DMICCDAT */ + 0x0000 /* 5:Default */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000000..96cf9fc9c8b0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec Device Tree Bindings + +maintainers: + - patches@opensource.cirrus.com + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt deleted file mode 100644 index f660d9bb0e69..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8731.txt +++ /dev/null @@ -1,27 +0,0 @@ -WM8731 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8731" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8731: codec@1a { - compatible = "wlf,wm8731"; - reg = <0x1a>; -}; - -Available audio endpoints for an audio-routing table: - * LOUT: Left Channel Line Output - * ROUT: Right Channel Line Output - * LHPOUT: Left Channel Headphone Output - * RHPOUT: Right Channel Headphone Output - * LLINEIN: Left Channel Line Input - * RLINEIN: Right Channel Line Input - * MICIN: Microphone Input diff --git a/Documentation/devicetree/bindings/sound/wm8750.txt b/Documentation/devicetree/bindings/sound/wm8750.txt deleted file mode 100644 index 682f221f6f38..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8750.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8750 and WM8987 audio CODECs - -These devices support both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8750" or "wlf,wm8987" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8750: codec@1a { - compatible = "wlf,wm8750"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8750.yaml b/Documentation/devicetree/bindings/sound/wm8750.yaml new file mode 100644 index 000000000000..24246ac7bbdf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8750.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wm8750.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8750 and WM8987 audio CODECs + +description: | + These devices support both I2C and SPI (configured with pin strapping + on the board). + +maintainers: + - Mark Brown <broonie@kernel.org> + +properties: + compatible: + enum: + - wlf,wm8750 + - wlf,wm8987 + + reg: + description: + The I2C address of the device for I2C, the chip select number for SPI + maxItems: 1 + +additionalProperties: false + +required: + - reg + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8750"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt deleted file mode 100644 index 6371c2434afe..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ /dev/null @@ -1,82 +0,0 @@ -WM8903 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8903" - - - reg : the I2C address of the device. - - - gpio-controller : Indicates this device is a GPIO controller. - - - #gpio-cells : Should be two. The first cell is the pin number and the - second cell is used to specify optional parameters (currently unused). - -Optional properties: - - - interrupts : The interrupt line the codec is connected to. - - - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the - default is 0. - - - micdet-delay : The debounce delay for microphone detection in mS. If - absent, the default is 100. - - - gpio-cfg : A list of GPIO configuration register values. The list must - be 5 entries long. If absent, no configuration of these registers is - performed. If any entry has the value 0xffffffff, that GPIO's - configuration will not be modified. - - - AVDD-supply : Analog power supply regulator on the AVDD pin. - - - CPVDD-supply : Charge pump supply regulator on the CPVDD pin. - - - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin. - - - DCVDD-supply : Digital core supply regulator for the DCVDD pin. - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - -Example: - -wm8903: codec@1a { - compatible = "wlf,wm8903"; - reg = <0x1a>; - interrupts = < 347 >; - - AVDD-supply = <&fooreg_a>; - CPVDD-supply = <&fooreg_b>; - DBVDD-supply = <&fooreg_c>; - DCVDC-supply = <&fooreg_d>; - - gpio-controller; - #gpio-cells = <2>; - - micdet-cfg = <0>; - micdet-delay = <100>; - gpio-cfg = < - 0x0600 /* DMIC_LR, output */ - 0x0680 /* DMIC_DAT, input */ - 0x0000 /* GPIO, output, low */ - 0x0200 /* Interrupt, output */ - 0x01a0 /* BCLK, input, active high */ - >; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt index 6d29ac3750ee..85d3b287108c 100644 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -21,6 +21,17 @@ Optional properties: enabled and disabled together with HP_L and HP_R pins in response to jack detect events. + - wlf,hp-cfg: A list of headphone jack detect configuration register values. + The list must be 3 entries long. + hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4). + hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2). + hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1). + + - wlf,gpio-cfg: A list of GPIO configuration register values. + The list must be 2 entries long. + gpio-cfg[0]: ALRCGPIO of R9 (Audio interface) + gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4). + Example: wm8960: codec@1a { diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt deleted file mode 100644 index dcfa9a3369fd..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8962.txt +++ /dev/null @@ -1,39 +0,0 @@ -WM8962 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8962" - - - reg : the I2C address of the device. - -Optional properties: - - spk-mono: This is a boolean property. If present, the SPK_MONO bit - of R51 (Class D Control 2) gets set, indicating that the speaker is - in mono mode. - - - mic-cfg : Default register value for R48 (Additional Control 4). - If absent, the default should be the register default. - - - gpio-cfg : A list of GPIO configuration register values. The list must - be 6 entries long. If absent, no configuration of these registers is - performed. And note that only the value within [0x0, 0xffff] is valid. - Any other value is regarded as setting the GPIO register by its reset - value 0x0. - -Example: - -wm8962: codec@1a { - compatible = "wlf,wm8962"; - reg = <0x1a>; - - gpio-cfg = < - 0x0000 /* 0:Default */ - 0x0000 /* 1:Default */ - 0x0013 /* 2:FN_DMICCLK */ - 0x0000 /* 3:Default */ - 0x8014 /* 4:FN_DMICCDAT */ - 0x0000 /* 5:Default */ - >; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index 68cccc4653ba..8fa947509c10 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -14,9 +14,15 @@ Required properties: - #gpio-cells : Must be 2. The first cell is the pin number and the second cell is used to specify optional parameters (currently unused). - - AVDD2-supply, DBVDD1-supply, DBVDD2-supply, DBVDD3-supply, CPVDD-supply, - SPKVDD1-supply, SPKVDD2-supply : power supplies for the device, as covered - in Documentation/devicetree/bindings/regulator/regulator.txt + - power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt, depending + on compatible: + - for wlf,wm1811 and wlf,wm8958: + AVDD1-supply, AVDD2-supply, DBVDD1-supply, DBVDD2-supply, DBVDD3-supply, + DCVDD-supply, CPVDD-supply, SPKVDD1-supply, SPKVDD2-supply + - for wlf,wm8994: + AVDD1-supply, AVDD2-supply, DBVDD-supply, DCVDD-supply, CPVDD-supply, + SPKVDD1-supply, SPKVDD2-supply Optional properties: @@ -62,6 +68,29 @@ Optional properties: - wlf,csnaddr-pd : If present enable the internal pull-down resistor on the CS/ADDR pin. +Pins on the device (for linking into audio routes): + + * IN1LN + * IN1LP + * IN2LN + * IN2LP:VXRN + * IN1RN + * IN1RP + * IN2RN + * IN2RP:VXRP + * SPKOUTLP + * SPKOUTLN + * SPKOUTRP + * SPKOUTRN + * HPOUT1L + * HPOUT1R + * HPOUT2P + * HPOUT2N + * LINEOUT1P + * LINEOUT1N + * LINEOUT2P + * LINEOUT2N + Example: wm8994: codec@1a { @@ -73,11 +102,11 @@ wm8994: codec@1a { lineout1-se; + AVDD1-supply = <®ulator>; AVDD2-supply = <®ulator>; CPVDD-supply = <®ulator>; - DBVDD1-supply = <®ulator>; - DBVDD2-supply = <®ulator>; - DBVDD3-supply = <®ulator>; + DBVDD-supply = <®ulator>; + DCVDD-supply = <®ulator>; SPKVDD1-supply = <®ulator>; SPKVDD2-supply = <®ulator>; }; diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000000..338e2a13c775 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex@perex.cz> + - Takashi Iwai <tiwai@suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/zte,tdm.txt b/Documentation/devicetree/bindings/sound/zte,tdm.txt deleted file mode 100644 index 2a07ca655264..000000000000 --- a/Documentation/devicetree/bindings/sound/zte,tdm.txt +++ /dev/null @@ -1,30 +0,0 @@ -ZTE TDM DAI driver - -Required properties: - -- compatible : should be one of the following. - * zte,zx296718-tdm -- reg : physical base address of the controller and length of memory mapped - region. -- clocks : Pairs of phandle and specifier referencing the controller's clocks. -- clock-names: "wclk" for the wclk. - "pclk" for the pclk. --#clock-cells: should be 1. -- zte,tdm-dma-sysctrl : Reference to the sysctrl controller controlling - the dma. includes: - phandle of sysctrl. - register offset in sysctrl for control dma. - mask of the register that be written to sysctrl. - -Example: - - tdm: tdm@1487000 { - compatible = "zte,zx296718-tdm"; - reg = <0x01487000 0x1000>; - clocks = <&audiocrm AUDIO_TDM_WCLK>, <&audiocrm AUDIO_TDM_PCLK>; - clock-names = "wclk", "pclk"; - #clock-cells = <1>; - pinctrl-names = "default"; - pinctrl-0 = <&tdm_global_pin>; - zte,tdm-dma-sysctrl = <&sysctrl 0x10c 4>; - }; diff --git a/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt b/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt deleted file mode 100644 index 41bb1040eb71..000000000000 --- a/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt +++ /dev/null @@ -1,24 +0,0 @@ -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 = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; - #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>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt b/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt deleted file mode 100644 index 3927251464f0..000000000000 --- a/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt +++ /dev/null @@ -1,45 +0,0 @@ -ZTE ZX296702 I2S controller - -Required properties: - - compatible : Must be one of: - "zte,zx296718-i2s", "zte,zx296702-i2s" - "zte,zx296702-i2s" - - reg : Must contain I2S core's registers location and length - - clocks : Pairs of phandle and specifier referencing the controller's clocks. - - clock-names: "wclk" for the wclk, "pclk" for the pclk to the I2S interface. - - dmas: Pairs of phandle and specifier for the DMA channel that is used by - the core. The core expects two dma channels for transmit. - - dma-names : Must be "tx" and "rx" - -For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties -please check: - * resource-names.txt - * clock/clock-bindings.txt - * dma/dma.txt - -Example: - i2s0: i2s@b005000 { - #sound-dai-cells = <0>; - compatible = "zte,zx296718-i2s", "zte,zx296702-i2s"; - reg = <0x0b005000 0x1000>; - clocks = <&audiocrm AUDIO_I2S0_WCLK>, <&audiocrm AUDIO_I2S0_PCLK>; - clock-names = "wclk", "pclk"; - interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dma 5>, <&dma 6>; - dma-names = "tx", "rx"; - }; - - sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "zx296702_snd"; - 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 = <&i2s0>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&acodec>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt b/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt deleted file mode 100644 index 09231d7586b2..000000000000 --- a/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt +++ /dev/null @@ -1,27 +0,0 @@ -ZTE ZX296702 SPDIF controller - -Required properties: - - compatible : Must be "zte,zx296702-spdif" - - reg : Must contain SPDIF core's registers location and length - - clocks : Pairs of phandle and specifier referencing the controller's clocks. - - clock-names: "tx" for the clock to the SPDIF interface. - - dmas: Pairs of phandle and specifier for the DMA channel that is used by - the core. The core expects one dma channel for transmit. - - dma-names : Must be "tx" - -For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties -please check: - * resource-names.txt - * clock/clock-bindings.txt - * dma/dma.txt - -Example: - spdif0: spdif0@b004000 { - compatible = "zte,zx296702-spdif"; - reg = <0x0b004000 0x1000>; - clocks = <&lsp0clk ZX296702_SPDIF0_DIV>; - clock-names = "tx"; - interrupts = <GIC_SPI 21 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dma 4>; - dma-names = "tx"; - }; |