diff options
Diffstat (limited to 'Documentation/devicetree/bindings/sound')
418 files changed, 22811 insertions, 6341 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..ea62e51aba90 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml @@ -0,0 +1,69 @@ +# 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.com> + +description: | + Analog Devices ADAU1372 four inputs and two outputs codec. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf + +allOf: + - $ref: dai-common.yaml# + +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 + +unevaluatedProperties: 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,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt deleted file mode 100644 index 1447dec28125..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ /dev/null @@ -1,32 +0,0 @@ -Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781 - -Required properties: - - - compatible: Should contain one of the following: - "adi,adau1361" - "adi,adau1461" - "adi,adau1761" - "adi,adau1961" - "adi,adau1381" - "adi,adau1781" - - - reg: The i2c address. Value depends on the state of ADDR0 - and ADDR1, as wired in hardware. - -Optional properties: - - clock-names: If provided must be "mclk". - - clocks: phandle + clock-specifiers for the clock that provides - the audio master clock for the device. - -Examples: -#include <dt-bindings/sound/adau17x1.h> - - i2c_bus { - adau1361@38 { - compatible = "adi,adau1761"; - reg = <0x38>; - - clock-names = "mclk"; - clocks = <&audio_clock>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.yaml b/Documentation/devicetree/bindings/sound/adi,adau17x1.yaml new file mode 100644 index 000000000000..8ef1e7f6ec91 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.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/adi,adau17x1.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781 Codec + +maintainers: + - Lars-Peter Clausen <lars@metafoo.de> + +properties: + compatible: + enum: + - adi,adau1361 + - adi,adau1381 + - adi,adau1461 + - adi,adau1761 + - adi,adau1781 + - adi,adau1961 + + reg: + maxItems: 1 + description: + The i2c address. Value depends on the state of ADDR0 and ADDR1, + as wired in hardware. + + clock-names: + const: mclk + + clocks: + items: + - description: provides the audio master clock for the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@38 { + compatible = "adi,adau1761"; + reg = <0x38>; + clock-names = "mclk"; + clocks = <&audio_clock>; + }; + }; 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..dba3023a45e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.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/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 + + 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 + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + spi { + #address-cells = <1>; + #size-cells = <0>; + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + reg = <0>; + spi-max-frequency = <600000>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + + adi,micbias = <3>; + }; + }; + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt deleted file mode 100644 index f144ee1abf85..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau7002.txt +++ /dev/null @@ -1,19 +0,0 @@ -Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter - -Required properties: - - - compatible: Must be "adi,adau7002" - -Optional properties: - - - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD - supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt - - If this property is not present it is assumed that the supply pin is - hardwired to always on. - -Example: - adau7002: pdm-to-i2s { - compatible = "adi,adau7002"; - IOVDD-supply = <&supply>; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.yaml b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml new file mode 100644 index 000000000000..fcca0fde7d86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,adau7002 + + IOVDD-supply: + description: + IOVDD power supply, if skipped then it is assumed that the supply pin is + hardwired to always on. + + wakeup-delay-ms: + description: + Delay after power up needed for device to settle. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + audio-codec { + compatible = "adi,adau7002"; + IOVDD-supply = <&pp1800_l15a>; + #sound-dai-cells = <0>; + wakeup-delay-ms = <80>; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index fb78967ee17b..12f60507aed7 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -15,6 +15,9 @@ description: | standalone mode. https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU7118.pdf +allOf: + - $ref: dai-common.yaml# + properties: compatible: enum: @@ -57,7 +60,7 @@ required: - iovdd-supply - dvdd-supply -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/adi,max98363.yaml b/Documentation/devicetree/bindings/sound/adi,max98363.yaml new file mode 100644 index 000000000000..c388cda56011 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98363.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98363.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98363 SoundWire Amplifier + +maintainers: + - Ryan Lee <ryans.lee@analog.com> + +description: + The MAX98363 is a SoundWire input Class D mono amplifier that + supports MIPI SoundWire v1.2-compatible digital interface for + audio and control data. + SoundWire peripheral device ID of MAX98363 is 0x3*019f836300 + where * is the peripheral device unique ID decoded from pin. + It supports up to 10 peripheral devices(0x0 to 0x9). + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw3019f836300 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + soundwire@3250000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x3250000 0x2000>; + + speaker@0,0 { + compatible = "sdw3019f836300"; + reg = <0 0>; + #sound-dai-cells = <0>; + sound-name-prefix = "Speaker Left"; + }; + + speaker@0,1 { + compatible = "sdw3019f836300"; + reg = <0 1>; + #sound-dai-cells = <0>; + sound-name-prefix = "Speaker Right"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,max98388.yaml b/Documentation/devicetree/bindings/sound/adi,max98388.yaml new file mode 100644 index 000000000000..93ccd5905736 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98388.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98388.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98388 Speaker Amplifier + +maintainers: + - Ryan Lee <ryans.lee@analog.com> + +description: + The MAX98388 is a mono Class-D speaker amplifier with I/V feedback. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,max98388 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + adi,vmon-slot-no: + description: slot number of the voltage feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 1 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V feedback data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98388: amplifier@39 { + compatible = "adi,max98388"; + reg = <0x39>; + #sound-dai-cells = <0>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml new file mode 100644 index 000000000000..bdc10d4204ec --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -0,0 +1,141 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98396.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98396 Speaker Amplifier + +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. + + avdd-supply: + description: A 1.8V supply that powers up the AVDD pin. + + dvdd-supply: + description: A 1.2V supply that powers up the DVDD pin. + + dvddio-supply: + description: A 1.2V or 1.8V supply that powers up the VDDIO pin. + + pvdd-supply: + description: A 3.0V to 20V supply that powers up the PVDD pin. + + vbat-supply: + description: A 3.3V to 5.5V supply that powers up the VBAT pin. + + 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: 1 + + adi,spkfb-slot-no: + description: slot number of speaker DSP monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 2 + + adi,bypass-slot-no: + description: + Selects the PCM data input channel that is routed to the speaker + audio processing bypass path. + $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 + + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-ms: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + enum: [64, 256, 1024, 4096] + default: 64 + + 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>; + dvdd-supply = <®ulator_1v2>; + dvddio-supply = <®ulator_1v8>; + avdd-supply = <®ulator_1v8>; + pvdd-supply = <®ulator_pvdd>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml b/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml new file mode 100644 index 000000000000..f3f32540779c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,ssm2518.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices SSM2518 audio amplifier + +maintainers: + - Lars-Peter Clausen <lars@metafoo.de> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,ssm2518 + + reg: + maxItems: 1 + description: | + I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + + gpios: + maxItems: 1 + description: | + GPIO connected to the nSD pin. If the property is not present + it is assumed that the nSD pin is hardwired to always on. + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml b/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml new file mode 100644 index 000000000000..144450df5869 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,ssm3515.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices SSM3515 Audio Amplifier + +maintainers: + - Martin PoviÅ¡er <povik+lin@cutebit.org> + +description: | + SSM3515 is a mono Class-D audio amplifier with digital input. + + https://www.analog.com/media/en/technical-documentation/data-sheets/SSM3515.pdf + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,ssm3515 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@14 { + compatible = "adi,ssm3515"; + reg = <0x14>; + #sound-dai-cells = <0>; + sound-name-prefix = "Left Tweeter"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4375.yaml b/Documentation/devicetree/bindings/sound/ak4375.yaml new file mode 100644 index 000000000000..587598e122c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4375.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4375 DAC and headphones amplifier + +maintainers: + - Vincent Knecht <vincent.knecht@mailoo.org> + +allOf: + - $ref: dai-common.yaml# + +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 + +unevaluatedProperties: 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 deleted file mode 100644 index e5820235e0d5..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ /dev/null @@ -1,23 +0,0 @@ -AK4458 audio DAC - -This device supports I2C mode. - -Required properties: - -- compatible : "asahi-kasei,ak4458" or "asahi-kasei,ak4497" -- reg : The I2C address of the device for I2C - -Optional properties: -- reset-gpios: A GPIO specifier for the power down & reset pin -- mute-gpios: A GPIO specifier for the soft mute pin - -Example: - -&i2c { - ak4458: dac@10 { - compatible = "asahi-kasei,ak4458"; - reg = <0x10>; - reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW> - mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH> - }; -}; 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..75e13414d6eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,59 @@ +# 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 + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + ports: + $ref: audio-graph-port.yaml#/definitions/ports + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: codec@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt deleted file mode 100644 index 58e48ee97175..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4642.txt +++ /dev/null @@ -1,37 +0,0 @@ -AK4642 I2C transmitter - -This device supports I2C mode only. - -Required properties: - - - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" - - reg : The chip select number on the I2C bus - -Optional properties: - - - #clock-cells : common clock binding; shall be set to 0 - - clocks : common clock binding; MCKI clock - - clock-frequency : common clock binding; frequency of MCKO - - clock-output-names : common clock binding; MCKO clock name - -Example 1: - -&i2c { - ak4648: ak4648@12 { - compatible = "asahi-kasei,ak4642"; - reg = <0x12>; - }; -}; - -Example 2: - -&i2c { - ak4643: codec@12 { - compatible = "asahi-kasei,ak4643"; - reg = <0x12>; - #clock-cells = <0>; - clocks = <&audio_clock>; - clock-frequency = <12288000>; - clock-output-names = "ak4643_mcko"; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml new file mode 100644 index 000000000000..437fe5d7cae1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: dai-common.yaml# + +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 + +unevaluatedProperties: 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 deleted file mode 100644 index 7d67ca6ced80..000000000000 --- a/Documentation/devicetree/bindings/sound/ak5558.txt +++ /dev/null @@ -1,22 +0,0 @@ -AK5558 8 channel differential 32-bit delta-sigma ADC - -This device supports I2C mode only. - -Required properties: - -- compatible : "asahi-kasei,ak5558" -- reg : The I2C address of the device. - -Optional properties: - -- reset-gpios: A GPIO specifier for the power down & reset pin. - -Example: - -&i2c { - ak5558: adc@10 { - compatible = "asahi-kasei,ak5558"; - reg = <0x10>; - reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt deleted file mode 100644 index ffd886d110bd..000000000000 --- a/Documentation/devicetree/bindings/sound/alc5632.txt +++ /dev/null @@ -1,43 +0,0 @@ -ALC5632 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "realtek,alc5632" - - - 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). - -Pins on the device (for linking into audio routes): - - * SPK_OUTP - * SPK_OUTN - * HP_OUT_L - * HP_OUT_R - * AUX_OUT_P - * AUX_OUT_N - * LINE_IN_L - * LINE_IN_R - * PHONE_P - * PHONE_N - * MIC1_P - * MIC1_N - * MIC2_P - * MIC2_N - * MICBIAS1 - * DMICDAT - -Example: - -alc5632: alc5632@1e { - compatible = "realtek,alc5632"; - reg = <0x1a>; - - gpio-controller; - #gpio-cells = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml index be390accdd07..78273647f766 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -4,7 +4,7 @@ $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 +title: Allwinner A10 Codec maintainers: - Chen-Yu Tsai <wens@csie.org> @@ -57,7 +57,7 @@ properties: 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 + $ref: /schemas/types.yaml#/definitions/non-unique-string-array minItems: 2 maxItems: 18 items: @@ -88,6 +88,7 @@ properties: description: Phandle to the codec analog controls in the PRCM allwinner,pa-gpios: + maxItems: 1 description: GPIO to enable the external amplifier required: @@ -101,6 +102,7 @@ required: - dma-names allOf: + - $ref: dai-common.yaml# - if: properties: compatible: @@ -227,7 +229,7 @@ allOf: - Mic - Speaker -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml index 112ae00d63c1..739114fb6549 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml @@ -4,7 +4,7 @@ $id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-i2s.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Allwinner A10 I2S Controller Device Tree Bindings +title: Allwinner A10 I2S Controller maintainers: - Chen-Yu Tsai <wens@csie.org> @@ -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 @@ -50,6 +61,7 @@ properties: maxItems: 1 allOf: + - $ref: dai-common.yaml# - if: properties: compatible: @@ -59,6 +71,8 @@ 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 +82,14 @@ 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 @@ -114,7 +129,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml index 444a432912bb..aa32dc950e72 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -4,7 +4,7 @@ $id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-spdif.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Allwinner A10 S/PDIF Controller Device Tree Bindings +title: Allwinner A10 S/PDIF Controller maintainers: - Chen-Yu Tsai <wens@csie.org> @@ -22,6 +22,7 @@ properties: - const: allwinner,sun6i-a31-spdif - const: allwinner,sun8i-h3-spdif - const: allwinner,sun50i-h6-spdif + - const: allwinner,sun50i-h616-spdif - items: - const: allwinner,sun8i-a83t-spdif - const: allwinner,sun8i-h3-spdif @@ -54,6 +55,7 @@ properties: maxItems: 1 allOf: + - $ref: dai-common.yaml# - if: properties: compatible: @@ -61,6 +63,8 @@ allOf: enum: - allwinner,sun6i-a31-spdif - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif + - allwinner,sun50i-h616-spdif then: required: @@ -72,7 +76,7 @@ allOf: contains: enum: - allwinner,sun8i-h3-spdif - - allwinner,sun50i-h6-spdif + - allwinner,sun50i-h616-spdif then: properties: @@ -104,7 +108,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml index 3b764415c9ab..5800de63fc84 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml @@ -4,7 +4,7 @@ $id: http://devicetree.org/schemas/sound/allwinner,sun50i-a64-codec-analog.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Allwinner A64 Analog Codec Device Tree Bindings +title: Allwinner A64 Analog Codec maintainers: - Chen-Yu Tsai <wens@csie.org> @@ -21,6 +21,11 @@ properties: description: Regulator for the headphone amplifier + allwinner,internal-bias-resistor: + description: + Enable the internal 2.2K bias resistor between HBIAS and MICDET pins + type: boolean + required: - compatible - reg diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..763b876047c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC + +maintainers: + - Ban Tao <fengzheng923@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - items: + - enum: + - allwinner,sun20i-d1-dmic + - const: allwinner,sun50i-h6-dmic + - const: allwinner,sun50i-h6-dmic + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml index 9718358826ab..1c21a1b390c7 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml @@ -4,7 +4,7 @@ $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 +title: Allwinner A23 Analog Codec maintainers: - Chen-Yu Tsai <wens@csie.org> @@ -12,12 +12,15 @@ maintainers: properties: compatible: - enum: + oneOf: # FIXME: This is documented in the PRCM binding, but needs to be # migrated here at some point # - allwinner,sun8i-a23-codec-analog - - allwinner,sun8i-h3-codec-analog - - allwinner,sun8i-v3s-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 diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml index 55d28268d2f4..63eadc4200ac 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml @@ -4,18 +4,29 @@ $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 +title: Allwinner A33 Codec maintainers: - Chen-Yu Tsai <wens@csie.org> - Maxime Ripard <mripard@kernel.org> +allOf: + - $ref: dai-common.yaml# + properties: "#sound-dai-cells": - const: 0 + 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: - const: allwinner,sun8i-a33-codec + oneOf: + - items: + - const: allwinner,sun50i-a64-codec + - const: allwinner,sun8i-a33-codec + - const: allwinner,sun8i-a33-codec reg: maxItems: 1 @@ -41,12 +52,12 @@ required: - clocks - clock-names -additionalProperties: false +unevaluatedProperties: false examples: - | audio-codec@1c22e00 { - #sound-dai-cells = <0>; + #sound-dai-cells = <1>; compatible = "allwinner,sun8i-a33-codec"; reg = <0x01c22e00 0x400>; interrupts = <0 29 4>; diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml index f9344adaf6c2..6350dfc0a926 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -9,6 +9,9 @@ title: Amlogic AIU audio output controller maintainers: - Jerome Brunet <jbrunet@baylibre.com> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: "^audio-controller@.*" @@ -19,12 +22,11 @@ properties: compatible: items: - enum: - - amlogic,aiu-gxbb - - amlogic,aiu-gxl - - amlogic,aiu-meson8 - - amlogic,aiu-meson8b - - const: - amlogic,aiu + - amlogic,aiu-gxbb + - amlogic,aiu-gxl + - amlogic,aiu-meson8 + - amlogic,aiu-meson8b + - const: amlogic,aiu clocks: items: @@ -66,6 +68,8 @@ properties: resets: maxItems: 1 + sound-name-prefix: true + required: - "#sound-dai-cells" - compatible @@ -76,6 +80,8 @@ required: - reg - resets +additionalProperties: false + examples: - | #include <dt-bindings/clock/gxbb-clkc.h> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt deleted file mode 100644 index fa4545ed81ca..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt +++ /dev/null @@ -1,34 +0,0 @@ -* Amlogic Audio FIFO controllers - -Required properties: -- compatible: 'amlogic,axg-toddr' or - 'amlogic,axg-toddr' or - 'amlogic,g12a-frddr' or - '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: 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: - -frddr_a: audio-controller@1c0 { - compatible = "amlogic,axg-frddr"; - reg = <0x0 0x1c0 0x0 0x1c>; - #sound-dai-cells = <0>; - 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-fifo.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml new file mode 100644 index 000000000000..b1b48d683101 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml @@ -0,0 +1,112 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-fifo.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AXG Audio FIFO controllers + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - enum: + - amlogic,axg-toddr + - amlogic,axg-frddr + - items: + - enum: + - amlogic,g12a-toddr + - amlogic,sm1-toddr + - const: amlogic,axg-toddr + - items: + - enum: + - amlogic,g12a-frddr + - amlogic,sm1-frddr + - const: amlogic,axg-frddr + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + + interrupts: + maxItems: 1 + + resets: + minItems: 1 + maxItems: 2 + + reset-names: + minItems: 1 + maxItems: 2 + + amlogic,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Size of the controller's fifo in bytes + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - interrupts + - resets + - amlogic,fifo-depth + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-toddr + - amlogic,sm1-toddr + - amlogic,g12a-frddr + - amlogic,sm1-frddr + + then: + properties: + resets: + minItems: 2 + reset-names: + items: + - const: arb + - const: rst + required: + - reset-names + + else: + properties: + resets: + maxItems: 1 + reset-names: + const: arb + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/reset/amlogic,meson-axg-audio-arb.h> + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + audio-controller@1c0 { + compatible = "amlogic,g12a-frddr", "amlogic,axg-frddr"; + reg = <0x1c0 0x1c>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; + interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; + resets = <&arb>, <&clkc_audio AUD_RESET_FRDDR_A>; + reset-names = "arb", "rst"; + amlogic,fifo-depth = <512>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt deleted file mode 100644 index 716878107a24..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt +++ /dev/null @@ -1,29 +0,0 @@ -* Amlogic Audio PDM input - -Required properties: -- compatible: 'amlogic,axg-pdm' or - '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. -- clock-names: should contain the following: - * "pclk" : peripheral clock. - * "dclk" : pdm digital clock - * "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 { - compatible = "amlogic,axg-pdm"; - reg = <0x0 0xff632000 0x0 0x34>; - #sound-dai-cells = <0>; - clocks = <&clkc_audio AUD_CLKID_PDM>, - <&clkc_audio AUD_CLKID_PDM_DCLK>, - <&clkc_audio AUD_CLKID_PDM_SYSCLK>; - clock-names = "pclk", "dclk", "sysclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml new file mode 100644 index 000000000000..df21dd72fc65 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.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/amlogic,axg-pdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG PDM input + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - items: + - enum: + - amlogic,g12a-pdm + - amlogic,sm1-pdm + - const: amlogic,axg-pdm + - const: amlogic,axg-pdm + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: PDM digital clock + - description: DSP system clock + + clock-names: + items: + - const: pclk + - const: dclk + - const: sysclk + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-pdm + - amlogic,sm1-pdm + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller@ff632000 { + compatible = "amlogic,axg-pdm"; + reg = <0xff632000 0x34>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_PDM>, + <&clkc_audio AUD_CLKID_PDM_DCLK>, + <&clkc_audio AUD_CLKID_PDM_SYSCLK>; + clock-names = "pclk", "dclk", "sysclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt deleted file mode 100644 index 80b411296480..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt +++ /dev/null @@ -1,124 +0,0 @@ -Amlogic AXG sound card: - -Required properties: - -- compatible: "amlogic,axg-sound-card" -- model : User specified audio sound card name, one string - -Optional properties: - -- audio-aux-devs : List of phandles pointing to auxiliary devices -- audio-widgets : Please refer to widgets.txt. -- audio-routing : A list of the connections between audio components. - -Subnodes: - -- dai-link: Container for dai-link level properties and the CODEC - sub-nodes. There should be at least one (and probably more) - subnode of this type. - -Required dai-link properties: - -- sound-dai: phandle and port of the CPU DAI. - -Required TDM Backend dai-link properties: -- dai-format : CPU/CODEC common audio format - -Optional TDM Backend dai-link properties: -- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks -- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks - When omitted, mask is assumed to have to no - slots. A valid must have at one slot, so at - least one these mask should be provided with - an enabled slot. -- dai-tdm-slot-num : Please refer to tdm-slot.txt. - If omitted, slot number is set to accommodate the largest - mask provided. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted. -- mclk-fs : Multiplication factor between stream rate and mclk - -Backend dai-link subnodes: - -- codec: 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 - -Required codec subnodes properties: - -- sound-dai: phandle and port of the CODEC DAI. - -Optional codec subnodes properties: - -- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt. -- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt. - -Example: - -sound { - compatible = "amlogic,axg-sound-card"; - model = "AXG-S420"; - audio-aux-devs = <&tdmin_a>, <&tdmout_c>; - audio-widgets = "Line", "Lineout", - "Line", "Linein", - "Speaker", "Speaker1 Left", - "Speaker", "Speaker1 Right"; - "Speaker", "Speaker2 Left", - "Speaker", "Speaker2 Right"; - audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", - "SPDIFOUT IN 0", "FRDDR_A OUT 3", - "TDM_C Playback", "TDMOUT_C OUT", - "TDMIN_A IN 2", "TDM_C Capture", - "TDMIN_A IN 5", "TDM_C Loopback", - "TODDR_A IN 0", "TDMIN_A OUT", - "Lineout", "Lineout AOUTL", - "Lineout", "Lineout AOUTR", - "Speaker1 Left", "SPK1 OUT_A", - "Speaker2 Left", "SPK2 OUT_A", - "Speaker1 Right", "SPK1 OUT_B", - "Speaker2 Right", "SPK2 OUT_B", - "Linein AINL", "Linein", - "Linein AINR", "Linein"; - - dai-link@0 { - sound-dai = <&frddr_a>; - }; - - dai-link@1 { - sound-dai = <&toddr_a>; - }; - - dai-link@2 { - sound-dai = <&tdmif_c>; - dai-format = "i2s"; - dai-tdm-slot-tx-mask-2 = <1 1>; - dai-tdm-slot-tx-mask-3 = <1 1>; - dai-tdm-slot-rx-mask-1 = <1 1>; - mclk-fs = <256>; - - codec@0 { - sound-dai = <&lineout>; - }; - - codec@1 { - sound-dai = <&speaker_amp1>; - }; - - codec@2 { - sound-dai = <&speaker_amp2>; - }; - - codec@3 { - sound-dai = <&linein>; - }; - - }; - - dai-link@3 { - sound-dai = <&spdifout>; - - codec { - sound-dai = <&spdif_dit>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml new file mode 100644 index 000000000000..5db718e4d0e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -0,0 +1,174 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AXG sound card + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: amlogic,axg-sound-card + + audio-aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: list of auxiliary devices + + audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + +patternProperties: + "^dai-link-[0-9]+$": + type: object + additionalProperties: false + description: + 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 ] + + dai-tdm-slot-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of slots in use. If omitted, slot number is set to + accommodate the largest mask provided. + maximum: 32 + + dai-tdm-slot-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Width in bits for each slot + enum: [ 8, 16, 20, 24, 32 ] + default: 32 + + mclk-fs: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Multiplication factor between the frame rate and master clock + rate + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + patternProperties: + "^dai-tdm-slot-(t|r)x-mask-[0-3]$": + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 32 + description: + Transmit and receive cpu slot masks of each TDM lane + When omitted, mask is assumed to have to no slots. A valid + interface must have at least one slot, so at least one these + mask should be provided with an enabled slot. + + "^codec(-[0-9]+)?$": + type: object + additionalProperties: false + description: + 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 + + patternProperties: + "^dai-tdm-slot-(t|r)x-mask$": + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 32 + description: Transmit and receive codec slot masks + + required: + - sound-dai + + required: + - sound-dai + +required: + - dai-link-0 + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "amlogic,axg-sound-card"; + model = "AXG-S420"; + audio-aux-devs = <&tdmin_a>, <&tdmout_c>; + audio-widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right", + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; + + dai-link-0 { + sound-dai = <&frddr_a>; + }; + + dai-link-1 { + sound-dai = <&toddr_a>; + }; + + dai-link-2 { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-2 = <1 1>; + dai-tdm-slot-tx-mask-3 = <1 1>; + dai-tdm-slot-rx-mask-1 = <1 1>; + mclk-fs = <256>; + + codec-0 { + sound-dai = <&lineout>; + }; + + codec-1 { + sound-dai = <&speaker_amp1>; + }; + + codec-2 { + sound-dai = <&speaker_amp2>; + }; + + codec-3 { + sound-dai = <&linein>; + }; + }; + + dai-link-3 { + sound-dai = <&spdifout>; + + codec { + sound-dai = <&spdif_dit>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt deleted file mode 100644 index df92a4ecf288..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt +++ /dev/null @@ -1,27 +0,0 @@ -* Amlogic Audio SPDIF Input - -Required properties: -- compatible: 'amlogic,axg-spdifin' or - '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: - * "pclk" : peripheral clock. - * "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 { - compatible = "amlogic,axg-spdifin"; - reg = <0x0 0x400 0x0 0x30>; - #sound-dai-cells = <0>; - interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>; - clocks = <&clkc_audio AUD_CLKID_SPDIFIN>, - <&clkc_audio AUD_CLKID_SPDIFIN_CLK>; - clock-names = "pclk", "refclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml new file mode 100644 index 000000000000..a0bd7a5fb9b3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifin.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG SPDIF Input + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - const: amlogic,axg-spdifin + - items: + - enum: + - amlogic,g12a-spdifin + - amlogic,sm1-spdifin + - const: amlogic,axg-spdifin + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: SPDIF input reference clock + + clock-names: + items: + - const: pclk + - const: refclk + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + - interrupts + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-spdifin + - amlogic,sm1-spdifin + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + audio-controller@400 { + compatible = "amlogic,axg-spdifin"; + reg = <0x400 0x30>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>; + clocks = <&clkc_audio AUD_CLKID_SPDIFIN>, + <&clkc_audio AUD_CLKID_SPDIFIN_CLK>; + clock-names = "pclk", "refclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt deleted file mode 100644 index 28381dd1f633..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt +++ /dev/null @@ -1,25 +0,0 @@ -* Amlogic Audio SPDIF Output - -Required properties: -- compatible: 'amlogic,axg-spdifout' or - '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 { - compatible = "amlogic,axg-spdifout"; - reg = <0x0 0x480 0x0 0x50>; - #sound-dai-cells = <0>; - clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, - <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; - clock-names = "pclk", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml new file mode 100644 index 000000000000..15be8dae9398 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.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/amlogic,axg-spdifout.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG SPDIF Output + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - const: amlogic,axg-spdifout + - items: + - enum: + - amlogic,g12a-spdifout + - amlogic,sm1-spdifout + - const: amlogic,axg-spdifout + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: SPDIF output master clock + + clock-names: + items: + - const: pclk + - const: mclk + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-spdifout + - amlogic,sm1-spdifout + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller@480 { + compatible = "amlogic,axg-spdifout"; + reg = <0x480 0x50>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, + <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; + clock-names = "pclk", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt deleted file mode 100644 index 5996c0cd89c2..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt +++ /dev/null @@ -1,36 +0,0 @@ -* Amlogic Audio TDM formatters - -Required properties: -- compatible: 'amlogic,axg-tdmin' or - 'amlogic,axg-tdmout' or - 'amlogic,g12a-tdmin' or - '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. -- clock-names: should contain the following: - * "pclk" : peripheral clock. - * "sclk" : bit clock. - * "sclk_sel" : bit clock input multiplexer. - * "lrclk" : sample clock - * "lrclk_sel": sample clock input multiplexer - -Optional property: -- resets: phandle to the dedicated reset line of the tdm formatter. - -Example of TDMOUT_A on the S905X2 SoC: - -tdmout_a: audio-controller@500 { - compatible = "amlogic,axg-tdmout"; - reg = <0x0 0x500 0x0 0x40>; - resets = <&clkc_audio AUD_RESET_TDMOUT_A>; - clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, - <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, - <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, - <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, - <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; - clock-names = "pclk", "sclk", "sclk_sel", - "lrclk", "lrclk_sel"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml new file mode 100644 index 000000000000..719ca8fc98c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-formatters.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG TDM formatters + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + enum: + - amlogic,g12a-tdmout + - amlogic,sm1-tdmout + - amlogic,axg-tdmout + - amlogic,g12a-tdmin + - amlogic,sm1-tdmin + - amlogic,axg-tdmin + + clocks: + items: + - description: Peripheral clock + - description: Bit clock + - description: Bit clock input multiplexer + - description: Sample clock + - description: Sample clock input multiplexer + + clock-names: + items: + - const: pclk + - const: sclk + - const: sclk_sel + - const: lrclk + - const: lrclk_sel + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + +allOf: + - $ref: component-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-tdmin + - amlogic,sm1-tdmin + - amlogic,g12a-tdmout + - amlogic,sm1-tdmout + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + audio-controller@500 { + compatible = "amlogic,g12a-tdmout"; + reg = <0x500 0x40>; + resets = <&clkc_audio AUD_RESET_TDMOUT_A>; + clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; + clock-names = "pclk", "sclk", "sclk_sel", + "lrclk", "lrclk_sel"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt deleted file mode 100644 index cabfb26a5f22..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt +++ /dev/null @@ -1,22 +0,0 @@ -* Amlogic Audio TDM Interfaces - -Required properties: -- compatible: 'amlogic,axg-tdm-iface' -- clocks: list of clock phandle, one for each entry clock-names. -- clock-names: should contain the following: - * "sclk" : bit clock. - * "lrclk": sample clock - * "mclk" : master clock - -> optional if the interface is in clock slave mode. -- #sound-dai-cells: must be 0. - -Example of TDM_A on the A113 SoC: - -tdmif_a: audio-controller@0 { - compatible = "amlogic,axg-tdm-iface"; - #sound-dai-cells = <0>; - clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>, - <&clkc_audio AUD_CLKID_MST_A_SCLK>, - <&clkc_audio AUD_CLKID_MST_A_LRCLK>; - clock-names = "mclk", "sclk", "lrclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml new file mode 100644 index 000000000000..45955d8a26d1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.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/amlogic,axg-tdm-iface.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio TDM Interfaces + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: amlogic,axg-tdm-iface + + "#sound-dai-cells": + const: 0 + + clocks: + minItems: 2 + items: + - description: Bit clock + - description: Sample clock + - description: Master clock # optional + + clock-names: + minItems: 2 + items: + - const: sclk + - const: lrclk + - const: mclk + +required: + - compatible + - "#sound-dai-cells" + - clocks + - clock-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller { + compatible = "amlogic,axg-tdm-iface"; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_MST_A_SCLK>, + <&clkc_audio AUD_CLKID_MST_A_LRCLK>, + <&clkc_audio AUD_CLKID_MST_A_MCLK>; + clock-names = "sclk", "lrclk", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml index 51a0c30e10f9..23f82bb89750 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -9,6 +9,9 @@ title: Amlogic G12a Internal DAC Control Glue maintainers: - Jerome Brunet <jbrunet@baylibre.com> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: "^audio-controller@.*" @@ -19,13 +22,11 @@ properties: compatible: oneOf: - items: - - const: - amlogic,g12a-toacodec + - const: amlogic,g12a-toacodec - items: - - enum: - - amlogic,sm1-toacodec - - const: - amlogic,g12a-toacodec + - enum: + - amlogic,sm1-toacodec + - const: amlogic,g12a-toacodec reg: maxItems: 1 @@ -33,12 +34,16 @@ properties: 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> diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index fb374c659be1..d4277d342e69 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic GX sound card maintainers: - Jerome Brunet <jbrunet@baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: items: @@ -18,14 +21,6 @@ properties: $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 @@ -33,13 +28,10 @@ properties: 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 + additionalProperties: false description: |- dai-link child nodes: Container for dai-link level properties and the CODEC sub-nodes. @@ -57,12 +49,13 @@ patternProperties: rate sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle of the CPU DAI patternProperties: - "^codec-[0-9]+$": + "^codec(-[0-9]+)?$": type: object + additionalProperties: false description: |- Codecs: dai-link representing backend links should have at least one subnode. @@ -71,7 +64,7 @@ patternProperties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle of the codec DAI required: @@ -84,6 +77,8 @@ required: - model - dai-link-0 +unevaluatedProperties: false + examples: - | sound { @@ -110,4 +105,3 @@ examples: }; }; }; - diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml index 04014e658c90..5f5cccdbeb34 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -9,6 +9,9 @@ title: Amlogic T9015 Internal Audio DAC maintainers: - Jerome Brunet <jbrunet@baylibre.com> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: "^audio-controller@.*" @@ -34,6 +37,12 @@ properties: resets: maxItems: 1 + AVDD-supply: + description: + Analogue power supply. + + sound-name-prefix: true + required: - "#sound-dai-cells" - compatible @@ -41,6 +50,9 @@ required: - clocks - clock-names - resets + - AVDD-supply + +additionalProperties: false examples: - | @@ -54,4 +66,5 @@ examples: clocks = <&clkc CLKID_AUDIO_CODEC>; clock-names = "pclk"; resets = <&reset RESET_AUDIO_CODEC>; + AVDD-supply = <&vddao_1v8>; }; diff --git a/Documentation/devicetree/bindings/sound/apple,mca.yaml b/Documentation/devicetree/bindings/sound/apple,mca.yaml new file mode 100644 index 000000000000..5c6ec08c7d24 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/apple,mca.yaml @@ -0,0 +1,135 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/apple,mca.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Apple MCA I2S transceiver + +description: | + MCA is an I2S transceiver peripheral found on M1 and other Apple chips. It is + composed of a number of identical clusters which can operate independently + or in an interlinked fashion. Up to 6 clusters have been seen on an MCA. + +maintainers: + - Martin PoviÅ¡er <povik+lin@cutebit.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + items: + - enum: + - apple,t6000-mca + - apple,t8103-mca + - apple,t8112-mca + - const: apple,mca + + reg: + items: + - description: Register region of the MCA clusters proper + - description: Register region of the DMA glue and its FIFOs + + interrupts: + minItems: 4 + maxItems: 6 + description: + One interrupt per each cluster + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + + dmas: + minItems: 16 + maxItems: 24 + description: + DMA channels corresponding to the SERDES units in the peripheral. They are + listed in groups of four per cluster, and within the group they are given + as associated to the TXA, RXA, TXB, RXB units. + + dma-names: + minItems: 16 + items: + - const: tx0a + - const: rx0a + - const: tx0b + - const: rx0b + - const: tx1a + - const: rx1a + - const: tx1b + - const: rx1b + - const: tx2a + - const: rx2a + - const: tx2b + - const: rx2b + - const: tx3a + - const: rx3a + - const: tx3b + - const: rx3b + - const: tx4a + - const: rx4a + - const: tx4b + - const: rx4b + - const: tx5a + - const: rx5a + - const: tx5b + - const: rx5b + description: | + Names for the DMA channels: 'tx'/'rx', then cluster number, then 'a'/'b' + based on the associated SERDES unit. + + clocks: + minItems: 4 + maxItems: 6 + description: + Clusters' input reference clock. + + resets: + maxItems: 1 + + power-domains: + minItems: 5 + maxItems: 7 + description: + First a general power domain for register access, then the power + domains of individual clusters for their operation. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - power-domains + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + mca: i2s@9b600000 { + compatible = "apple,t6000-mca", "apple,mca"; + reg = <0x9b600000 0x10000>, + <0x9b200000 0x20000>; + + clocks = <&nco 0>, <&nco 1>, <&nco 2>, <&nco 3>; + power-domains = <&ps_audio_p>, <&ps_mca0>, <&ps_mca1>, + <&ps_mca2>, <&ps_mca3>; + dmas = <&admac 0>, <&admac 1>, <&admac 2>, <&admac 3>, + <&admac 4>, <&admac 5>, <&admac 6>, <&admac 7>, + <&admac 8>, <&admac 9>, <&admac 10>, <&admac 11>, + <&admac 12>, <&admac 13>, <&admac 14>, <&admac 15>; + dma-names = "tx0a", "rx0a", "tx0b", "rx0b", + "tx1a", "rx1a", "tx1b", "rx1b", + "tx2a", "rx2a", "tx2b", "rx2b", + "tx3a", "rx3a", "tx3b", "rx3b"; + + #sound-dai-cells = <1>; + }; 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 17530120ccfc..000000000000 --- a/Documentation/devicetree/bindings/sound/arndale.txt +++ /dev/null @@ -1,25 +0,0 @@ -Audio Binding for Arndale boards - -Required properties: -- compatible : Can be one of the following: - "samsung,arndale-rt5631", - "samsung,arndale-wm1811" - -- 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/asahi-kasei,ak4458.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml new file mode 100644 index 000000000000..4477f84b7acc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4458.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4458 audio DAC + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4458 + - asahi-kasei,ak4497 + + reg: + maxItems: 1 + + avdd-supply: + description: Analog power supply + + dvdd-supply: + description: Digital power supply + + reset-gpios: + maxItems: 1 + + mute-gpios: + maxItems: 1 + description: + GPIO used to mute all the outputs + + dsd-path: + description: Select DSD input pins for ak4497 + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: "select #16, #17, #19 pins" + - const: 1 + description: "select #3, #4, #5 pins" + +required: + - compatible + - reg + +allOf: + - if: + properties: + compatible: + contains: + const: asahi-kasei,ak4458 + + then: + properties: + dsd-path: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@10 { + compatible = "asahi-kasei,ak4458"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak5558.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak5558.yaml new file mode 100644 index 000000000000..d3d494ae8abf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak5558.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak5558.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK5558 8 channel differential 32-bit delta-sigma ADC + +maintainers: + - Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp> + - Mihai Serban <mihai.serban@nxp.com> + +properties: + compatible: + enum: + - asahi-kasei,ak5552 + - asahi-kasei,ak5558 + + reg: + maxItems: 1 + + avdd-supply: + description: A 1.8V supply that powers up the AVDD pin. + + dvdd-supply: + description: A 1.2V supply that powers up the DVDD pin. + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak5558: codec@10 { + compatible = "asahi-kasei,ak5558"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml b/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml new file mode 100644 index 000000000000..89a67f8e33b6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,asoc-wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel wm8904 audio codec complex + +maintainers: + - Dharma Balasubiramani <dharma.b@microchip.com> + +description: + The ASoC audio complex configuration for Atmel with WM8904 audio codec. + +properties: + compatible: + const: atmel,asoc-wm8904 + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8731 audio codec. + + atmel,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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + - Mic + # WM8904 CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - MICBIAS + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml b/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml new file mode 100644 index 000000000000..33717b728f63 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sam9x5-wm8731-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9x5ek wm8731 audio complex + +maintainers: + - Dharma Balasubiramani <dharma.b@microchip.com> + +description: + The audio complex configuration for Atmel at91sam9x5ek with WM8731 audio codec. + +properties: + compatible: + const: atmel,sam9x5-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8731 audio codec. + + atmel,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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + + # CODEC Pins + - LOUT + - ROUT + - LHPOUT + - RHPOUT + - LLINEIN + - RLINEIN + - MICIN + +required: + - compatible + - atmel,model + - atmel,ssc-controller + - atmel,audio-codec + - atmel,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,sam9x5-wm8731-audio"; + + atmel,model = "wm8731 @ AT91SAM9X5EK"; + + atmel,audio-routing = + "Headphone Jack", "RHPOUT", + "Headphone Jack", "LHPOUT", + "LLINEIN", "Line In Jack", + "RLINEIN", "Line In Jack"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml new file mode 100644 index 000000000000..ae3162fcfe02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-classd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel ClassD Amplifier + +maintainers: + - Nicolas Ferre <nicolas.ferre@microchip.com> + - Alexandre Belloni <alexandre.belloni@bootlin.com> + - Claudiu Beznea <claudiu.beznea@microchip.com> + +description: + The Audio Class D Amplifier (CLASSD) is a digital input, Pulse Width + Modulated (PWM) output stereo Class D amplifier. + +properties: + compatible: + oneOf: + - items: + - const: atmel,sama5d2-classd + - items: + - const: microchip,sam9x7-classd + - const: atmel,sama5d2-classd + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + const: tx + + clocks: + maxItems: 2 + + clock-names: + items: + - const: pclk + - const: gclk + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + default: CLASSD + description: The user-visible name of this sound complex. + + atmel,pwm-type: + $ref: /schemas/types.yaml#/definitions/string + enum: + - single + - diff + default: single + description: PWM modulation type. + + atmel,non-overlap-time: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 5 + - 10 + - 15 + - 20 + default: 10 + description: + Set non-overlapping time, the unit is nanosecond(ns). + Non-overlapping will be disabled if not specified. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clock-names + - clocks + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + classd: sound@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>; + clock-names = "pclk", "gclk"; + assigned-clocks = <&classd_gclk>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml new file mode 100644 index 000000000000..0cd1ff89baed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel I2S controller + +maintainers: + - Nicolas Ferre <nicolas.ferre@microchip.com> + - Alexandre Belloni <alexandre.belloni@bootlin.com> + - Claudiu Beznea <claudiu.beznea@microchip.com> + +description: + Atmel I2S (Inter-IC Sound Controller) bus is the standard + interface for connecting audio devices, such as audio codecs. + +properties: + compatible: + const: atmel,sama5d2-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + - description: Generated clock (Optional) + - description: I2S mux clock (Optional). Set + with gclk when Master Mode is required. + minItems: 1 + + clock-names: + items: + - const: pclk + - const: gclk + - const: muxclk + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f8050000 { + compatible = "atmel,sama5d2-i2s"; + reg = <0xf8050000 0x300>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(31))>, + <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(32))>; + dma-names = "tx", "rx"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s0_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml new file mode 100644 index 000000000000..f320b561f24c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-pdmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel PDMIC decoder + +maintainers: + - Claudiu Beznea <claudiu.beznea@microchip.com> + +description: + Atmel Pulse Density Modulation Interface Controller + (PDMIC) peripheral is a mono PDM decoder module + that decodes an incoming PDM sample stream. + +properties: + compatible: + const: atmel,sama5d2-pdmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: peripheral clock + - description: generated clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + maxItems: 1 + + dma-names: + const: rx + + atmel,mic-min-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The minimal frequency that the microphone supports. + + atmel,mic-max-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The maximal frequency that the microphone supports. + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + default: PDMIC + description: The user-visible name of this sound card. + + atmel,mic-offset: + $ref: /schemas/types.yaml#/definitions/int32 + default: 0 + description: The offset that should be added. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clock-names + - clocks + - atmel,mic-min-freq + - atmel,mic-max-freq + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + pdmic: sound@f8018000 { + compatible = "atmel,sama5d2-pdmic"; + reg = <0xf8018000 0x124>; + interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(50))>; + dma-names = "rx"; + clocks = <&pdmic_clk>, <&pdmic_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pdmic_default>; + atmel,model = "PDMIC@sama5d2_xplained"; + atmel,mic-min-freq = <1000000>; + atmel,mic-max-freq = <3246000>; + atmel,mic-offset = <0x0>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt deleted file mode 100644 index 898551076382..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-classd.txt +++ /dev/null @@ -1,55 +0,0 @@ -* Atmel ClassD driver under ALSA SoC architecture - -Required properties: -- compatible - Should be "atmel,sama5d2-classd". -- reg - Should contain ClassD registers location and length. -- interrupts - Should contain the IRQ line for the ClassD. -- dmas - One DMA specifiers as described in atmel-dma.txt and dma.txt files. -- dma-names - Must be "tx". -- clock-names - Tuple listing input clock names. - Required elements: "pclk" and "gclk". -- clocks - Please refer to clock-bindings.txt. -- assigned-clocks - Should be <&classd_gclk>. - -Optional properties: -- pinctrl-names, pinctrl-0 - Please refer to pinctrl-bindings.txt. -- atmel,model - The user-visible name of this sound complex. - The default value is "CLASSD". -- atmel,pwm-type - PWM modulation type, "single" or "diff". - The default value is "single". -- atmel,non-overlap-time - Set non-overlapping time, the unit is nanosecond(ns). - There are four values, - <5>, <10>, <15>, <20>, the default value is <10>. - Non-overlapping will be disabled if not specified. - -Example: -classd: classd@fc048000 { - compatible = "atmel,sama5d2-classd"; - reg = <0xfc048000 0x100>; - interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; - dmas = <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) - | AT91_XDMAC_DT_PERID(47))>; - dma-names = "tx"; - clocks = <&classd_clk>, <&classd_gclk>; - clock-names = "pclk", "gclk"; - assigned-clocks = <&classd_gclk>; - - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_classd_default>; - atmel,model = "classd @ SAMA5D2-Xplained"; - atmel,pwm-type = "diff"; - atmel,non-overlap-time = <10>; -}; diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt deleted file mode 100644 index 40549f496a81..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-i2s.txt +++ /dev/null @@ -1,46 +0,0 @@ -* Atmel I2S controller - -Required properties: -- compatible: Should be "atmel,sama5d2-i2s". -- 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: Two dmas have to be defined, "tx" and "rx". - This IP also supports one shared channel for both rx and tx; - if this mode is used, one "rx-tx" name must be used. -- 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). - - "muxclk" (I2S mux 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 and the I2S - mux clock are optional and should only be set together, when Master Mode - is required. - -Example: - - i2s@f8050000 { - compatible = "atmel,sama5d2-i2s"; - reg = <0xf8050000 0x300>; - interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; - dmas = <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(31))>, - <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(32))>; - dma-names = "tx", "rx"; - clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; - clock-names = "pclk", "gclk", "muxclk"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s0_default>; - }; diff --git a/Documentation/devicetree/bindings/sound/atmel-pdmic.txt b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt deleted file mode 100644 index e0875f17c229..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-pdmic.txt +++ /dev/null @@ -1,55 +0,0 @@ -* Atmel PDMIC driver under ALSA SoC architecture - -Required properties: -- compatible - Should be "atmel,sama5d2-pdmic". -- reg - Should contain PDMIC registers location and length. -- interrupts - Should contain the IRQ line for the PDMIC. -- dmas - One DMA specifiers as described in atmel-dma.txt and dma.txt files. -- dma-names - Must be "rx". -- clock-names - Required elements: - - "pclk" peripheral clock - - "gclk" generated clock -- clocks - Must contain an entry for each required entry in clock-names. - Please refer to clock-bindings.txt. -- atmel,mic-min-freq - The minimal frequency that the micphone supports. -- atmel,mic-max-freq - The maximal frequency that the micphone supports. - -Optional properties: -- pinctrl-names, pinctrl-0 - Please refer to pinctrl-bindings.txt. -- atmel,model - The user-visible name of this sound card. - The default value is "PDMIC". -- atmel,mic-offset - The offset that should be added. - The range is from -32768 to 32767. - The default value is 0. - -Example: - pdmic@f8018000 { - compatible = "atmel,sama5d2-pdmic"; - reg = <0xf8018000 0x124>; - interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; - dmas = <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) - | AT91_XDMAC_DT_PERID(50))>; - dma-names = "rx"; - clocks = <&pdmic_clk>, <&pdmic_gclk>; - clock-names = "pclk", "gclk"; - - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pdmic_default>; - atmel,model = "PDMIC @ sama5d2_xplained"; - atmel,mic-min-freq = <1000000>; - atmel,mic-max-freq = <3246000>; - atmel,mic-offset = <0x0>; - }; diff --git a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt deleted file mode 100644 index 0720857089a7..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt +++ /dev/null @@ -1,35 +0,0 @@ -* Atmel at91sam9x5ek wm8731 audio complex - -Required properties: - - compatible: "atmel,sam9x5-wm8731-audio" - - atmel,model: The user-visible name of this sound complex. - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8731 audio codec - - atmel,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. - -Available audio endpoints for the audio-routing table: - -Board connectors: - * Headphone Jack - * Line In Jack - -wm8731 pins: -cf Documentation/devicetree/bindings/sound/wm8731.txt - -Example: -sound { - compatible = "atmel,sam9x5-wm8731-audio"; - - atmel,model = "wm8731 @ AT91SAM9X5EK"; - - atmel,audio-routing = - "Headphone Jack", "RHPOUT", - "Headphone Jack", "LHPOUT", - "LLINEIN", "Line In Jack", - "RLINEIN", "Line In Jack"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8731>; -}; diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt deleted file mode 100644 index 8bbe50c884b6..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt +++ /dev/null @@ -1,55 +0,0 @@ -Atmel ASoC driver with wm8904 audio codec complex - -Required properties: - - compatible: "atmel,asoc-wm8904" - - atmel,model: The user-visible name of this sound complex. - - atmel,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 WM8904's pins, and the jacks on the board: - - WM8904 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - - Board connectors: - - * Headphone Jack - * Line In Jack - * Mic - - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8904 audio codec - -Optional properties: - - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt - -Example: -sound { - compatible = "atmel,asoc-wm8904"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pck0_as_mck>; - - atmel,model = "wm8904 @ AT91SAM9N12EK"; - - atmel,audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "IN2L", "Line In Jack", - "IN2R", "Line In Jack", - "Mic", "MICBIAS", - "IN1L", "Mic"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8904>; -}; 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..274092ef36c5 --- /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 + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: /schemas/sound/audio-graph.yaml# + +properties: + compatible: + enum: + - audio-graph-card + - audio-graph-scu-card + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "audio-graph-card"; + + dais = <&cpu_port_a>; + }; + + cpu { + /* + * dai-controller own settings + */ + + port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + dai-format = "left_j"; + }; + }; + }; + + codec { + /* + * codec own settings + */ + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml new file mode 100644 index 000000000000..d3ce4de449d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-card2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card2 + +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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + multi: + type: object + description: Multi-CPU/Codec node + dpcm: + type: object + description: DPCM node + codec2codec: + type: object + description: Codec to Codec node + +required: + - compatible + - links + +additionalProperties: false + +... diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml new file mode 100644 index 000000000000..28b27e7e45de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -0,0 +1,122 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-port.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card 'port' + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +definitions: + port-base: + allOf: + - $ref: /schemas/graph.yaml#/$defs/port-base + - $ref: /schemas/sound/dai-params.yaml# + properties: + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + playback-only: + description: port connection used only for playback + $ref: /schemas/types.yaml#/definitions/flag + capture-only: + description: port connection used only for capture + $ref: /schemas/types.yaml#/definitions/flag + + endpoint-base: + allOf: + - $ref: /schemas/graph.yaml#/$defs/endpoint-base + - $ref: /schemas/sound/dai-params.yaml# + properties: + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + 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 + clocks: + description: Indicates system clock + maxItems: 1 + system-clock-frequency: + $ref: simple-card.yaml#/definitions/system-clock-frequency + system-clock-direction-out: + $ref: simple-card.yaml#/definitions/system-clock-direction-out + system-clock-fixed: + $ref: simple-card.yaml#/definitions/system-clock-fixed + + dai-format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai-tdm-slot-num: + description: Number of slots in use. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: Width in bits for each slot. + $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 + + ports: + $ref: "#/definitions/port-base" + unevaluatedProperties: false + patternProperties: + "^port(@[0-9a-f]+)?$": + $ref: "#/definitions/port-base" + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: "#/definitions/endpoint-base" + unevaluatedProperties: false + +allOf: + - $ref: "#/definitions/port-base" + +patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: "#/definitions/endpoint-base" + unevaluatedProperties: false + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml new file mode 100644 index 000000000000..71f52f7e55f6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: /schemas/sound/dai-params.yaml# + +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. + Each entry is a pair of strings, the first being the type of + widget ("Microphone", "Line", "Headphone", "Speaker"), the + second being the machine specific name for the widget. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pa-gpios: + maxItems: 1 + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + +required: + - dais + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml b/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml new file mode 100644 index 000000000000..d3cc1ea4a175 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-iio-aux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio IIO auxiliary + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: + Auxiliary device based on Industrial I/O device channels + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: audio-iio-aux + + io-channels: + description: + Industrial I/O device channels used + + io-channel-names: + description: + Industrial I/O channel names related to io-channels. + These names are used to provides sound controls, widgets and routes names. + + snd-control-invert-range: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + A list of 0/1 flags defining whether or not the related channel is + inverted + items: + enum: [0, 1] + default: 0 + description: | + Invert the sound control value compared to the IIO channel raw value. + - 1: The related sound control value is inverted meaning that the + minimum sound control value correspond to the maximum IIO channel + raw value and the maximum sound control value correspond to the + minimum IIO channel raw value. + - 0: The related sound control value is not inverted meaning that the + minimum (resp maximum) sound control value correspond to the + minimum (resp maximum) IIO channel raw value. + +required: + - compatible + - io-channels + - io-channel-names + +unevaluatedProperties: false + +examples: + - | + iio-aux { + compatible = "audio-iio-aux"; + io-channels = <&iio 0>, <&iio 1>, <&iio 2>, <&iio 3>; + io-channel-names = "CH0", "CH1", "CH2", "CH3"; + /* Invert CH1 and CH2 */ + snd-control-invert-range = <0 1 1 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml new file mode 100644 index 000000000000..bc6c6b172238 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: dai-common.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) + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 7 + + sound-name-prefix: true + +required: + - compatible + - mode-gpios + - awinic,mode + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + audio-amplifier { + compatible = "awinic,aw8738"; + mode-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + awinic,mode = <5>; + sound-name-prefix = "Speaker Amp"; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw87390.yaml b/Documentation/devicetree/bindings/sound/awinic,aw87390.yaml new file mode 100644 index 000000000000..ba9d8767c5d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw87390.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw87390.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic Aw87390 Audio Amplifier + +maintainers: + - Weidong Wang <wangweidong.a@awinic.com> + +description: + The awinic aw87390 is specifically designed to improve + the musical output dynamic range, enhance the overall + sound quallity, which is a new high efficiency, low + noise, constant large volume, 6th Smart K audio amplifier. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: awinic,aw87390 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + awinic,audio-channel: + description: + It is used to distinguish multiple PA devices, so that different + configurations can be loaded to different PA devices + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + +required: + - compatible + - reg + - "#sound-dai-cells" + - awinic,audio-channel + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@58 { + compatible = "awinic,aw87390"; + reg = <0x58>; + #sound-dai-cells = <0>; + awinic,audio-channel = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml new file mode 100644 index 000000000000..ac5f2e0f42cb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw88395.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic AW88395 Smart Audio Amplifier + +maintainers: + - Weidong Wang <wangweidong.a@awinic.com> + +description: + The Awinic AW88395 is an I2S/TDM input, high efficiency + digital Smart K audio amplifier with an integrated 10.25V + smart boost convert. + +properties: + compatible: + enum: + - awinic,aw88395 + - awinic,aw88261 + - awinic,aw88399 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + reset-gpios: + maxItems: 1 + + awinic,audio-channel: + description: + It is used to distinguish multiple PA devices, so that different + configurations can be loaded to different PA devices + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + + awinic,sync-flag: + description: + Flag bit used to keep the phase synchronized in the case of multiple PA + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + - '#sound-dai-cells' + - awinic,audio-channel + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - awinic,aw88261 + then: + properties: + reset-gpios: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@34 { + compatible = "awinic,aw88395"; + reg = <0x34>; + #sound-dai-cells = <0>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + awinic,audio-channel = <0>; + awinic,sync-flag; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt index 9d049d4bfd58..b6cc5f6f78c2 100644 --- a/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt +++ b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt @@ -29,7 +29,7 @@ The schematics explaining the gpios are as follows: IN2 +---o--+------------+--o---+ OUT2 loop2 relays -The 'loop1' gpio pin controlls two relays, which are either in loop position, +The 'loop1' gpio pin controls two relays, which are either in loop position, meaning that input and output are directly connected, or they are in mixer position, meaning that the signal is passed through the 'Sum' mixer. Similarly for 'loop2'. 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..14dea1feefc5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -0,0 +1,209 @@ +# 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 + + interrupts: + 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. + Shared boost allows two amplifiers to share a single boost circuit by + communicating on the MDSYNC bus. The active amplifier controls the boost + circuit using combined data from both amplifiers. GPIO1 should be + configured for Sync when shared boost is used. Shared boost is not + compatible with External boost. Active amplifier requires + boost-peak-milliamp, boost-ind-nanohenry and boost-cap-microfarad. + 0 = Internal Boost + 1 = External Boost + 2 = Shared Boost Active + 3 = Shared Boost Passive + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 + + 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: + - $ref: dai-common.yaml# + - if: + properties: + cirrus,boost-type: + const: 0 + then: + required: + - cirrus,boost-peak-milliamp + - cirrus,boost-ind-nanohenry + - cirrus,boost-cap-microfarad + else: + if: + properties: + cirrus,boost-type: + const: 1 + then: + required: + - cirrus,gpio1-output-enable + - cirrus,gpio1-src-select + properties: + cirrus,boost-peak-milliamp: false + cirrus,boost-ind-nanohenry: false + cirrus,boost-cap-microfarad: false + cirrus,gpio1-src-select: + enum: [1] + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l41: speaker-amp@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 GPIO_ACTIVE_HIGH>; + + cirrus,boost-type = <0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml new file mode 100644 index 000000000000..70f6c62aedca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml @@ -0,0 +1,159 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,cs35l45 + + reg: + maxItems: 1 + + interrupts: + 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. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 + default: 2 + +patternProperties: + "^cirrus,gpio-ctrl[1-3]$": + description: + GPIO pins configuration. + type: object + additionalProperties: false + properties: + gpio-dir: + description: + GPIO pin direction. Valid only when 'gpio-ctrl' is 1 + 0 = Output + 1 = Input + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 1 + gpio-lvl: + description: + GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 + 0 = Low + 1 = High + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 0 + gpio-op-cfg: + description: + GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 + 0 = CMOS + 1 = Open Drain + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 0 + gpio-pol: + description: + GPIO output polarity select. Valid only when 'gpio-ctrl' is 1 + and 'gpio-dir' is 0 + 0 = Non-inverted, Active High + 1 = Inverted, Active Low + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 0 + gpio-ctrl: + description: + Defines the function of the GPIO pin. + GPIO1 + 0 = High impedance input + 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + 2 = Pin acts as MDSYNC, direction controlled by MDSYNC + 3-7 = Reserved + GPIO2 + 0 = High impedance input + 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + 2 = Pin acts as open drain INT + 3 = Reserved + 4 = Pin acts as push-pull output INT. Active low. + 5 = Pin acts as push-pull output INT. Active high. + 6,7 = Reserved + GPIO3 + 0 = High impedance input + 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + 2-7 = Reserved + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + default: 0 +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/cs35l45.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l45: cs35l45@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l45"; + reg = <2>; + spi-max-frequency = <5000000>; + vdd-a-supply = <&dummy_vreg>; + vdd-batt-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + cirrus,asp-sdout-hiz-ctrl = <(CS35L45_ASP_TX_HIZ_UNUSED | + CS35L45_ASP_TX_HIZ_DISABLED)>; + cirrus,gpio-ctrl1 { + gpio-ctrl = <0x2>; + }; + cirrus,gpio-ctrl2 { + gpio-ctrl = <0x2>; + }; + cirrus,gpio-ctrl3 { + gpio-ctrl = <0x1>; + gpio-dir = <0x1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml new file mode 100644 index 000000000000..156560b2a980 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$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..af599d8735e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml @@ -0,0 +1,226 @@ +# 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 + - cirrus,cs42l83 + + 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. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 + + cirrus,hs-bias-sense-disable: + description: | + If present the HSBIAS sense is disabled. Configures HSBIAS output + current sense through the external 2.21-k resistor. HSBIAS_SENSE + is a hardware feature to reduce the potential pop noise when the + headset plug is removed slowly. But on some platforms ESD voltage + will affect it causing plug detection to fail, especially with CTIA + headset type. For different hardware setups, a designer might want + to tweak default behavior. + type: boolean + +required: + - compatible + - reg + - VP-supply + - VCP-supply + - VD_FILT-supply + - VL-supply + - VA-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs42l42.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l42: cs42l42@48 { + compatible = "cirrus,cs42l42"; + reg = <0x48>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD_FILT-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + + reset-gpios = <&axi_gpio_0 1 0>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + + cirrus,ts-inv = <CS42L42_TS_INV_DIS>; + cirrus,ts-dbnc-rise = <CS42L42_TS_DBNCE_1000>; + cirrus,ts-dbnc-fall = <CS42L42_TS_DBNCE_0>; + cirrus,btn-det-init-dbnce = <100>; + cirrus,btn-det-event-dbnce = <10>; + cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>; + cirrus,hs-bias-ramp-rate = <CS42L42_HSBIAS_RAMP_SLOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml new file mode 100644 index 000000000000..7f9d8c7a635a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml @@ -0,0 +1,313 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l43.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS42L43 Audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: | + The CS42L43 is an audio CODEC with integrated MIPI SoundWire interface + (Version 1.2.1 compliant), I2C, SPI, and I2S/TDM interfaces designed + for portable applications. It provides a high dynamic range, stereo + DAC for headphone output, two integrated Class D amplifiers for + loudspeakers, and two ADCs for wired headset microphone input or + stereo line input. PDM inputs are provided for digital microphones. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,cs42l43 + + reg: + maxItems: 1 + + vdd-p-supply: + description: + Power supply for the high voltage interface. + + vdd-a-supply: + description: + Power supply for internal analog circuits. + + vdd-d-supply: + description: + Power supply for internal digital circuits. Can be internally supplied. + + vdd-io-supply: + description: + Power supply for external interface and internal digital logic. + + vdd-cp-supply: + description: + Power supply for the amplifier 3 and 4 charge pump. + + vdd-amp-supply: + description: + Power supply for amplifier 1 and 2. + + reset-gpios: + maxItems: 1 + + interrupt-controller: true + + "#interrupt-cells": + const: 2 + + interrupts: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + clocks: + items: + - description: Synchronous audio clock provided on mclk_in. + + clock-names: + const: mclk + + cirrus,bias-low: + type: boolean + description: + Select a 1.8V headset micbias rather than 2.8V. + + cirrus,bias-sense-microamp: + description: + Current at which the headset micbias sense clamp will engage, 0 to + disable. + enum: [ 0, 14, 24, 43, 52, 61, 71, 90, 99 ] + default: 14 + + cirrus,bias-ramp-ms: + description: + Time in milliseconds the hardware allows for the headset micbias to + ramp up. + enum: [ 10, 40, 90, 170 ] + default: 170 + + cirrus,detect-us: + description: + Time in microseconds the type detection will run for. Long values will + cause more audible effects, but give more accurate detection. + enum: [ 20, 100, 1000, 10000, 50000, 75000, 100000, 200000 ] + default: 1000 + + cirrus,button-automute: + type: boolean + description: + Enable the hardware automuting of decimator 1 when a headset button is + pressed. + + cirrus,buttons-ohms: + description: + Impedance in Ohms for each headset button, these should be listed in + ascending order. + minItems: 1 + maxItems: 6 + + cirrus,tip-debounce-ms: + description: + Software debounce on tip sense triggering in milliseconds. + default: 0 + + cirrus,tip-invert: + type: boolean + description: + Indicates tip detect polarity, inverted implies open-circuit whilst the + jack is inserted. + + cirrus,tip-disable-pullup: + type: boolean + description: + Indicates if the internal pullup on the tip detect should be disabled. + + cirrus,tip-fall-db-ms: + description: + Time in milliseconds a falling edge on the tip detect should be hardware + debounced for. Note the falling edge is considered after the invert. + enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ] + default: 500 + + cirrus,tip-rise-db-ms: + description: + Time in milliseconds a rising edge on the tip detect should be hardware + debounced for. Note the rising edge is considered after the invert. + enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ] + default: 500 + + cirrus,use-ring-sense: + type: boolean + description: + Indicates if the ring sense should be used. + + cirrus,ring-invert: + type: boolean + description: + Indicates ring detect polarity, inverted implies open-circuit whilst the + jack is inserted. + + cirrus,ring-disable-pullup: + type: boolean + description: + Indicates if the internal pullup on the ring detect should be disabled. + + cirrus,ring-fall-db-ms: + description: + Time in milliseconds a falling edge on the ring detect should be hardware + debounced for. Note the falling edge is considered after the invert. + enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ] + default: 500 + + cirrus,ring-rise-db-ms: + description: + Time in milliseconds a rising edge on the ring detect should be hardware + debounced for. Note the rising edge is considered after the invert. + enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ] + default: 500 + + pinctrl: + type: object + $ref: /schemas/pinctrl/pinctrl.yaml# + additionalProperties: false + + properties: + gpio-controller: true + + "#gpio-cells": + const: 2 + + gpio-ranges: + items: + - description: A phandle to the CODEC pinctrl node + minimum: 0 + - const: 0 + - const: 0 + - const: 3 + + patternProperties: + "-state$": + oneOf: + - $ref: "#/$defs/cirrus-cs42l43-state" + - patternProperties: + "-pins$": + $ref: "#/$defs/cirrus-cs42l43-state" + additionalProperties: false + + spi: + type: object + $ref: /schemas/spi/spi-controller.yaml# + unevaluatedProperties: false + +$defs: + cirrus-cs42l43-state: + type: object + + allOf: + - $ref: /schemas/pinctrl/pincfg-node.yaml# + - $ref: /schemas/pinctrl/pinmux-node.yaml# + + oneOf: + - required: [ groups ] + - required: [ pins ] + + additionalProperties: false + + properties: + groups: + enum: [ gpio1, gpio2, gpio3, asp, pdmout2, pdmout1, i2c, spi ] + + pins: + enum: [ gpio1, gpio2, gpio3, + asp_dout, asp_fsync, asp_bclk, + pdmout2_clk, pdmout2_data, pdmout1_clk, pdmout1_data, + i2c_sda, i2c_scl, + spi_miso, spi_sck, spi_ssb ] + + function: + enum: [ gpio, spdif, irq, mic-shutter, spk-shutter ] + + drive-strength: + description: Set drive strength in mA + enum: [ 1, 2, 4, 8, 9, 10, 12, 16 ] + + input-debounce: + description: Set input debounce in uS + enum: [ 0, 85 ] + +required: + - compatible + - reg + - vdd-p-supply + - vdd-a-supply + - vdd-io-supply + - vdd-cp-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l43: codec@1a { + compatible = "cirrus,cs42l43"; + reg = <0x1a>; + + vdd-p-supply = <&vdd5v0>; + vdd-a-supply = <&vdd1v8>; + vdd-io-supply = <&vdd1v8>; + vdd-cp-supply = <&vdd1v8>; + vdd-amp-supply = <&vdd5v0>; + + reset-gpios = <&gpio 0>; + + interrupt-controller; + #interrupt-cells = <2>; + interrupt-parent = <&gpio>; + interrupts = <56 IRQ_TYPE_LEVEL_LOW>; + + #sound-dai-cells = <1>; + + clocks = <&clks 0>; + clock-names = "mclk"; + + cs42l43_pins: pinctrl { + gpio-controller; + #gpio-cells = <2>; + gpio-ranges = <&cs42l43_pins 0 0 3>; + + pinctrl-names = "default"; + pinctrl-0 = <&pinsettings>; + + pinsettings: default-state { + shutter-pins { + groups = "gpio3"; + function = "mic-shutter"; + }; + }; + }; + + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs-gpios = <&cs42l43_pins 1 0>; + + sensor@0 { + compatible = "bosch,bme680"; + reg = <0>; + spi-max-frequency = <1400000>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml index 83f44f07ac3f..f7bafbd4f1c2 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -4,14 +4,17 @@ $id: http://devicetree.org/schemas/sound/cirrus,cs42l51.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: CS42L51 audio codec DT bindings +title: CS42L51 audio codec maintainers: - - Olivier Moysan <olivier.moysan@st.com> + - Olivier Moysan <olivier.moysan@foss.st.com> + +allOf: + - $ref: dai-common.yaml# properties: compatible: - const: cirrus,cs42l51 + const: cirrus,cs42l51 reg: maxItems: 1 @@ -41,11 +44,17 @@ properties: VAHP-supply: description: phandle to voltage regulator of headphone + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - reg - "#sound-dai-cells" +unevaluatedProperties: false + examples: - | #include <dt-bindings/gpio/gpio.h> @@ -64,6 +73,13 @@ examples: VA-supply = <®_audio>; VAHP-supply = <®_audio>; reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; + + /* assume audio-graph */ + port { + cpu_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; }; }; ... diff --git a/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml b/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml new file mode 100644 index 000000000000..453d493c941f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,ep9301-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus EP93xx I2S Controller + +description: | + The I2S controller is used to stream serial audio data between the external + I2S CODECs’, ADCs/DACs, and the ARM Core. The controller supports I2S, Left- + and Right-Justified DSP formats. + +maintainers: + - Alexander Sverdlin <alexander.sverdlin@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: cirrus,ep9301-i2s + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 3 + maxItems: 3 + + clock-names: + items: + - const: mclk + - const: sclk + - const: lrclk + +required: + - compatible + - '#sound-dai-cells' + - reg + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + i2s: i2s@80820000 { + compatible = "cirrus,ep9301-i2s"; + #sound-dai-cells = <0>; + reg = <0x80820000 0x100>; + interrupt-parent = <&vic1>; + interrupts = <28>; + clocks = <&syscon 29>, + <&syscon 30>, + <&syscon 31>; + clock-names = "mclk", "sclk", "lrclk"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml index cea612d3d4a7..52f024f5302a 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml @@ -24,6 +24,9 @@ description: | This binding must be part of the Lochnagar MFD binding: [1] ../mfd/cirrus,lochnagar.yaml +allOf: + - $ref: dai-common.yaml# + properties: compatible: enum: @@ -49,4 +52,4 @@ required: - clocks - clock-names -additionalProperties: false +unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/cirrus,madera.yaml b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml index c4cd58b5acd4..014d4eaa8793 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,madera.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml @@ -22,6 +22,9 @@ description: | The properties are all contained in the parent MFD node. +allOf: + - $ref: dai-common.yaml# + properties: '#sound-dai-cells': description: @@ -111,3 +114,5 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 2 maxItems: 2 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/component-common.yaml b/Documentation/devicetree/bindings/sound/component-common.yaml new file mode 100644 index 000000000000..37766c5f3974 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/component-common.yaml @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/component-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Component Common Properties + +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/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt index 7915897f8a81..e84f30c5c39b 100644 --- a/Documentation/devicetree/bindings/sound/cs35l35.txt +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -110,7 +110,7 @@ Optional Monitor Signal Format sub-node: See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet - -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formatting on the I2S Port. Each of the 3 8 bit values in the array contain the settings for depth, location, and frame. diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt index 912bd162b477..d34117b8558e 100644 --- a/Documentation/devicetree/bindings/sound/cs35l36.txt +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -33,7 +33,7 @@ Optional properties: one amplifier in the system. If more than one it is best to Hi-Z the ASP port to prevent bus contention on the output signal - - cirrus,boost-ctl-select : Boost conerter control source selection. + - cirrus,boost-ctl-select : Boost converter control source selection. Selects the source of the BST_CTL target VBST voltage for the boost converter to generate. 0x00 - Control Port Value 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/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt index 12b4aa8ef0db..c1d5c8ad1a36 100644 --- a/Documentation/devicetree/bindings/sound/cs4341.txt +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -9,7 +9,7 @@ Required properties: number for SPI. For required properties on I2C-bus, please consult -Documentation/devicetree/bindings/i2c/i2c.txt +dtschema schemas/i2c/i2c-controller.yaml For required properties on SPI-bus, please consult Documentation/devicetree/bindings/spi/spi-bus.txt diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt index 4dbfb8274cd9..dc256adb35a2 100644 --- a/Documentation/devicetree/bindings/sound/cs53l30.txt +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -30,7 +30,7 @@ Optional properties: * frame using two different ways: * 1) Normal I2S mode on two data pins -- each SDOUT * carries 2-channel data in the same time. - * 2) TDM mode on one signle data pin -- SDOUT1 carries + * 2) TDM mode on one single data pin -- SDOUT1 carries * 4-channel data per frame. Example: diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt deleted file mode 100644 index add1caf26ac2..000000000000 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ /dev/null @@ -1,112 +0,0 @@ -Dialog Semiconductor DA7219 Audio Codec bindings - -DA7219 is an audio codec with advanced accessory detect features. - -====== - -Required properties: -- compatible : Should be "dlg,da7219" -- reg: Specifies the I2C slave address - -- interrupts : IRQ line info for DA7219. - (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for - further information relating to interrupt properties) - -- VDD-supply: VDD power supply for the device -- VDDMIC-supply: VDDMIC power supply for the device -- VDDIO-supply: VDDIO power supply for the device - (See Documentation/devicetree/bindings/regulator/regulator.txt for further - information relating to regulators) - -Optional properties: -- interrupt-names : Name associated with interrupt line. Should be "wakeup" if - interrupt is to be used to wake system, otherwise "irq" should be used. -- wakeup-source: Flag to indicate this device can wake system (suspend/resume). - -- #clock-cells : Should be set to '<1>', two clock sources provided; -- clock-output-names : Names given for DAI clock outputs (WCLK & BCLK); - -- clocks : phandle and clock specifier for codec MCLK. -- clock-names : Clock name string for 'clocks' attribute, should be "mclk". - -- dlg,micbias-lvl : Voltage (mV) for Mic Bias - [<1600>, <1800>, <2000>, <2200>, <2400>, <2600>] -- dlg,mic-amp-in-sel : Mic input source type - ["diff", "se_p", "se_n"] - -Deprecated properties: -- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine - (LDO unavailable in production HW so property no longer required). - -====== - -Child node - 'da7219_aad': - -Optional properties: -- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). - [<2800>, <2900>] -- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) -- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) - [<2>, <5>, <10>, <50>, <100>, <200>, <500>] -- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) - [<200>, <500>, <750>, <1000>] -- dlg,jack-ins-deb : Debounce time for jack insertion (ms) - [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] -- dlg,jack-det-rate: Jack type detection latency (3/4 pole) - ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] -- dlg,jack-rem-deb : Debounce time for jack removal (ms) - [<1>, <5>, <10>, <20>] -- dlg,a-d-btn-thr : Impedance threshold between buttons A and D - [0x0 - 0xFF] -- dlg,d-b-btn-thr : Impedance threshold between buttons D and B - [0x0 - 0xFF] -- dlg,b-c-btn-thr : Impedance threshold between buttons B and C - [0x0 - 0xFF] -- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic - [0x0 - 0xFF] -- dlg,btn-avg : Number of 8-bit readings for averaged button measurement - [<1>, <2>, <4>, <8>] -- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement - [<1>, <2>, <4>, <8>] - -====== - -Example: - - codec: da7219@1a { - compatible = "dlg,da7219"; - reg = <0x1a>; - - interrupt-parent = <&gpio6>; - interrupts = <11 IRQ_TYPE_LEVEL_LOW>; - - VDD-supply = <®_audio>; - VDDMIC-supply = <®_audio>; - VDDIO-supply = <®_audio>; - - #clock-cells = <1>; - clock-output-names = "dai-wclk", "dai-bclk"; - - clocks = <&clks 201>; - clock-names = "mclk"; - - dlg,ldo-lvl = <1200>; - dlg,micbias-lvl = <2600>; - dlg,mic-amp-in-sel = "diff"; - - da7219_aad { - dlg,btn-cfg = <50>; - dlg,mic-det-thr = <500>; - dlg,jack-ins-deb = <20>; - dlg,jack-det-rate = "32ms_64ms"; - dlg,jack-rem-deb = <1>; - - dlg,a-d-btn-thr = <0xa>; - dlg,d-b-btn-thr = <0x16>; - dlg,b-c-btn-thr = <0x21>; - dlg,c-mic-btn-thr = <0x3E>; - - dlg,btn-avg = <4>; - dlg,adc-1bit-rpt = <1>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/da9055.txt b/Documentation/devicetree/bindings/sound/da9055.txt index ed1b7cc6f249..75c6338b6ae2 100644 --- a/Documentation/devicetree/bindings/sound/da9055.txt +++ b/Documentation/devicetree/bindings/sound/da9055.txt @@ -2,7 +2,7 @@ DA9055 provides Audio CODEC support (I2C only). -The Audio CODEC device in DA9055 has it's own I2C address which is configurable, +The Audio CODEC device in DA9055 has its own I2C address which is configurable, so the device is instantiated separately from the PMIC (MFD) device. For details on accompanying PMIC I2C device, see the following: diff --git a/Documentation/devicetree/bindings/sound/dai-common.yaml b/Documentation/devicetree/bindings/sound/dai-common.yaml new file mode 100644 index 000000000000..1aed2f0f1775 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dai-common.yaml @@ -0,0 +1,18 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dai-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface Common Properties + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: component-common.yaml# + +properties: + '#sound-dai-cells': true + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/dai-params.yaml b/Documentation/devicetree/bindings/sound/dai-params.yaml new file mode 100644 index 000000000000..cd8508175564 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dai-params.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dai-params.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface (DAI) Stream Parameters + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +properties: + convert-channels: + description: Number of audio channels used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 32 + + convert-sample-format: + description: Audio sample format used by DAI + $ref: /schemas/types.yaml#/definitions/string + enum: + - s8 + - s16_le + - s24_le + - s24_3le + - s32_le + + convert-rate: + description: Audio sample rate used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 8000 + maximum: 192000 + +additionalProperties: true 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..7735e08d35ba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,202 @@ +# 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: + - $ref: dai-common.yaml# + - if: + properties: + opmode: + enum: + - 0 + + then: + required: + - tdm-slots + +unevaluatedProperties: 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/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt deleted file mode 100644 index 6a536d570e29..000000000000 --- a/Documentation/devicetree/bindings/sound/designware-i2s.txt +++ /dev/null @@ -1,35 +0,0 @@ -DesignWare I2S controller - -Required properties: - - compatible : Must be "snps,designware-i2s" - - reg : Must contain the I2S core's registers location and length - - clocks : Pairs of phandle and specifier referencing the controller's - clocks. The controller expects one clock: the clock used as the sampling - rate reference clock sample. - - clock-names : "i2sclk" for the sample rate reference clock. - - dmas: Pairs of phandle and specifier for the DMA channels that are used by - the core. The core expects one or two dma channels: one for transmit and - one for receive. - - dma-names : "tx" for the transmit channel, "rx" for the receive channel. - -Optional properties: - - interrupts: The interrupt line number for the I2S controller. Add this - parameter if the I2S controller that you are using does not support DMA. - -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: - - soc_i2s: i2s@7ff90000 { - compatible = "snps,designware-i2s"; - reg = <0x0 0x7ff90000 0x0 0x1000>; - clocks = <&scpi_i2sclk 0>; - clock-names = "i2sclk"; - #sound-dai-cells = <0>; - dmas = <&dma0 5>; - dma-names = "tx"; - }; diff --git a/Documentation/devicetree/bindings/sound/dialog,da7219.yaml b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml new file mode 100644 index 000000000000..19137abdba3e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml @@ -0,0 +1,238 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7219 Audio Codec + +maintainers: + - David Rau <David.Rau.opensource@dm.renesas.com> + +description: + The DA7219 is an ultra low-power audio codec with + in-built advanced accessory detection (AAD) for mobile + computing and accessory applications, which supports + sample rates up to 96 kHz at 24-bit resolution. + +properties: + compatible: + const: dlg,da7219 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + VDD-supply: + description: + VDD power supply for the device. + + VDDMIC-supply: + description: + VDDMIC power supply for the device. + + VDDIO-supply: + description: + VDDIO power supply for the device. + + interrupt-names: + description: + Should be "wakeup" if interrupt is to be used to wake system, + otherwise "irq" should be used. + enum: + - wakeup + - irq + + wakeup-source: + type: boolean + description: + Flag to indicate this device can wake system (suspend/resume). + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: + Name given for DAI WCLK and BCLK outputs. + + clocks: + maxItems: 1 + description: + phandle and clock specifier for codec MCLK. + + clock-names: + const: mclk + + dlg,micbias-lvl: + enum: [1600, 1800, 2000, 2200, 2400, 2600] + description: + Voltage (mV) for Mic Bias. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-amp-in-sel: + enum: [diff, se_p, se_n] + description: + Mic input source type. + + diff - Differential. + + se_p - MIC_P. + Positive differential analog microphone input. + + se_n - MIC_N. + Negative differential analog microphone input. + $ref: /schemas/types.yaml#/definitions/string + + da7219_aad: + type: object + additionalProperties: false + description: + Configuration of advanced accessory detection. + properties: + dlg,micbias-pulse-lvl: + enum: [2800, 2900] + description: + Mic bias higher voltage pulse level (mV). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,micbias-pulse-time: + description: + Mic bias higher voltage pulse duration (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + + dlg,btn-cfg: + enum: [2, 5, 10, 50, 100, 200, 500] + description: + Periodic button press measurements for 4-pole jack (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-det-thr: + enum: [200, 500, 750, 1000] + description: + Impedance threshold for mic detection measurement (Ohms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-deb: + enum: [5, 10, 20, 50, 100, 200, 500, 1000] + description: + Debounce time for jack insertion (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-det-pty: + enum: [low, high] + description: + Polarity for jack insertion detection. + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-det-rate: + enum: ["32_64", "64_128", "128_256", "256_512"] + description: + Jack type (3/4 pole) detection latency (ms). + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-rem-deb: + enum: [1, 5, 10, 20] + description: + Debounce time for jack removal (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,a-d-btn-thr: + description: + Impedance threshold between buttons A and D. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,d-b-btn-thr: + description: + Impedance threshold between buttons D and B. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,b-c-btn-thr: + description: + Impedance threshold between buttons B and C. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,c-mic-btn-thr: + description: + Impedance threshold between button C and Mic. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,btn-avg: + enum: [1, 2, 4, 8] + description: + Number of 8-bit readings for averaged button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,adc-1bit-rpt: + enum: [1, 2, 4, 8] + description: + Repeat count for 1-bit button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - interrupts + - VDD-supply + - VDDMIC-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <&vdd_reg>; + VDDMIC-supply = <&vddmic_reg>; + VDDIO-supply = <&vddio_reg>; + + #clock-cells = <1>; + clock-output-names = "da7219-dai-wclk", "da7219-dai-bclk"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-ins-det-pty = "low"; + dlg,jack-det-rate = "32_64"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/dmic-codec.yaml b/Documentation/devicetree/bindings/sound/dmic-codec.yaml new file mode 100644 index 000000000000..59ef0cf6b6e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dmic-codec.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dmic-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Generic PDM Digital microphone (DMIC) codec + +maintainers: + - Arnaud Pouliquen <arnaud.pouliquen@foss.st.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: dmic-codec + + '#sound-dai-cells': + const: 0 + + dmicen-gpios: + description: GPIO specifier for DMIC to control start and stop + maxItems: 1 + + num-channels: + description: Number of microphones on this DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 8 + default: 8 + + modeswitch-delay-ms: + description: Delay (in ms) to complete DMIC mode switch + + wakeup-delay-ms: + description: Delay (in ms) after enabling the DMIC + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + dmic { + compatible = "dmic-codec"; + dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; + num-channels = <1>; + wakeup-delay-ms = <50>; + modeswitch-delay-ms = <35>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt deleted file mode 100644 index 32e871037269..000000000000 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ /dev/null @@ -1,22 +0,0 @@ -Device-Tree bindings for Digital microphone (DMIC) codec - -This device support generic PDM digital microphone. - -Required properties: - - compatible: should be "dmic-codec". - -Optional properties: - - dmicen-gpios: GPIO specifier for dmic to control start and stop - - num-channels: Number of microphones on this DAI - - wakeup-delay-ms: Delay (in ms) after enabling the DMIC - - modeswitch-delay-ms: Delay (in ms) to complete DMIC mode switch - -Example node: - - dmic_codec: dmic@0 { - compatible = "dmic-codec"; - dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; - num-channels = <1>; - wakeup-delay-ms <50>; - modeswitch-delay-ms <35>; - }; diff --git a/Documentation/devicetree/bindings/sound/es8328.txt b/Documentation/devicetree/bindings/sound/es8328.txt deleted file mode 100644 index 33fbf058c997..000000000000 --- a/Documentation/devicetree/bindings/sound/es8328.txt +++ /dev/null @@ -1,38 +0,0 @@ -Everest ES8328 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : Should be "everest,es8328" or "everest,es8388" - - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V - - AVDD-supply : Regulator providing analog supply voltage 3.3V - - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V - - IPVDD-supply : Regulator providing analog output voltage 3.3V - - clocks : A 22.5792 or 11.2896 MHz clock - - reg : the I2C address of the device for I2C, the chip select number for SPI - -Pins on the device (for linking into audio routes): - - * LOUT1 - * LOUT2 - * ROUT1 - * ROUT2 - * LINPUT1 - * RINPUT1 - * LINPUT2 - * RINPUT2 - * Mic Bias - - -Example: - -codec: es8328@11 { - compatible = "everest,es8328"; - DVDD-supply = <®_3p3v>; - AVDD-supply = <®_3p3v>; - PVDD-supply = <®_3p3v>; - HPVDD-supply = <®_3p3v>; - clocks = <&clks 169>; - reg = <0x11>; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt deleted file mode 100644 index 1bf03c5f2af4..000000000000 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ /dev/null @@ -1,23 +0,0 @@ -Everest ES8316 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : should be "everest,es8316" - - reg : the I2C address of the device for I2C - -Optional properties: - - - clocks : a list of phandle, should contain entries for clock-names - - clock-names : should include as follows: - "mclk" : master clock (MCLK) of the device - -Example: - -es8316: codec@11 { - compatible = "everest,es8316"; - reg = <0x11>; - clocks = <&clks 10>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000000..b6079b3c440d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.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/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> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100644 index 000000000000..8c82d47375ec --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua@everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + everest,jack-pol: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the button on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x00 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [00]; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8328.yaml b/Documentation/devicetree/bindings/sound/everest,es8328.yaml new file mode 100644 index 000000000000..a0f4670fa38c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8328.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8328.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8328 audio CODEC + +description: + Everest Audio Codec, which can be connected via I2C or SPI. + Pins on the device (for linking into audio routes) are + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * LINPUT1 + * RINPUT1 + * LINPUT2 + * RINPUT2 + * Mic Bias + +maintainers: + - David Yang <yangxiaohua@everest-semi.com> + +properties: + compatible: + enum: + - everest,es8328 + - everest,es8388 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: A 22.5792 or 11.2896 MHz clock + + DVDD-supply: + description: Regulator providing digital core supply voltage 1.8 - 3.6V + + AVDD-supply: + description: Regulator providing analog supply voltage 3.3V + + PVDD-supply: + description: Regulator providing digital IO supply voltage 1.8 - 3.6V + + HPVDD-supply: + description: Regulator providing analog output voltage 3.3V + +required: + - compatible + - clocks + - DVDD-supply + - AVDD-supply + - PVDD-supply + - HPVDD-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8328: codec@11 { + compatible = "everest,es8328"; + reg = <0x11>; + AVDD-supply = <®_3p3v>; + DVDD-supply = <®_3p3v>; + HPVDD-supply = <®_3p3v>; + PVDD-supply = <®_3p3v>; + clocks = <&clks 169>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt deleted file mode 100644 index 998b4c8a7f78..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ /dev/null @@ -1,80 +0,0 @@ -Freescale Asynchronous Sample Rate Converter (ASRC) Controller - -The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a -signal associated with an input clock into a signal associated with a different -output clock. The driver currently works as a Front End of DPCM with other Back -Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support -three substreams within totally 10 channels. - -Required properties: - - - 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. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Contains the following entries - "mem" Peripheral access clock to access registers. - "ipg" Peripheral clock to driver module. - "asrck_<0-f>" Clock sources for input and output clock. - "spba" The spba clock is required when ASRC 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. - - - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. - - - 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 { - compatible = "fsl,imx53-asrc"; - reg = <0x02034000 0x4000>; - interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&clks 107>, <&clks 107>, <&clks 0>, - <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, - <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, - <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, - <&clks 107>, <&clks 0>, <&clks 0>; - clock-names = "mem", "ipg", "asrck0", - "asrck_1", "asrck_2", "asrck_3", "asrck_4", - "asrck_5", "asrck_6", "asrck_7", "asrck_8", - "asrck_9", "asrck_a", "asrck_b", "asrck_c", - "asrck_d", "asrck_e", "asrck_f"; - dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, - <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; - dma-names = "rxa", "rxb", "rxc", - "txa", "txb", "txc"; - fsl,asrc-rate = <48000>; - fsl,asrc-width = <16>; -}; 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 index 32d547af9ce7..0782f3f9947f 100644 --- a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -14,7 +14,13 @@ properties: pattern: "^easrc@.*" compatible: - const: fsl,imx8mn-easrc + oneOf: + - enum: + - fsl,imx8mn-easrc + - items: + - enum: + - fsl,imx8mp-easrc + - const: fsl,imx8mn-easrc reg: maxItems: 1 @@ -45,8 +51,8 @@ properties: - const: ctx3_tx firmware-name: - $ref: /schemas/types.yaml#/definitions/string - const: imx/easrc/easrc-imx8mn.bin + items: + - const: imx/easrc/easrc-imx8mn.bin description: The coefficient table for the filters fsl,asrc-rate: @@ -74,6 +80,8 @@ required: - fsl,asrc-rate - fsl,asrc-format +additionalProperties: false + examples: - | #include <dt-bindings/clock/imx8mn-clock.h> diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt index 0a2480aeecf0..90112ca1ff42 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -44,7 +44,7 @@ Required properties: - fsl,esai-synchronous: This is a boolean property. If present, indicating that ESAI would work in the synchronous mode, which means all the settings for Receiving would be - duplicated from Transmition related registers. + duplicated from Transmission related registers. Optional properties: diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml new file mode 100644 index 000000000000..bfef2fcb75b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml @@ -0,0 +1,162 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx-asrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Asynchronous Sample Rate Converter (ASRC) Controller + +description: + The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of + a signal associated with an input clock into a signal associated with a + different output clock. The driver currently works as a Front End of DPCM + with other Back Ends Audio controller such as ESAI, SSI and SAI. It has + three pairs to support three substreams within totally 10 channels. + +maintainers: + - Shawn Guo <shawnguo@kernel.org> + - Sascha Hauer <s.hauer@pengutronix.de> + +properties: + compatible: + oneOf: + - enum: + - fsl,imx35-asrc + - fsl,imx53-asrc + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + - items: + - enum: + - fsl,imx6sx-asrc + - fsl,imx6ul-asrc + - const: fsl,imx53-asrc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 6 + + dma-names: + items: + - const: rxa + - const: rxb + - const: rxc + - const: txa + - const: txb + - const: txc + + clocks: + maxItems: 19 + + clock-names: + items: + - const: mem + - const: ipg + - const: asrck_0 + - const: asrck_1 + - const: asrck_2 + - const: asrck_3 + - const: asrck_4 + - const: asrck_5 + - const: asrck_6 + - const: asrck_7 + - const: asrck_8 + - const: asrck_9 + - const: asrck_a + - const: asrck_b + - const: asrck_c + - const: asrck_d + - const: asrck_e + - const: asrck_f + - const: spba + + fsl,asrc-rate: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The mutual sample rate used by DPCM Back Ends + + fsl,asrc-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The mutual sample width used by DPCM Back Ends + enum: [16, 24] + + fsl,asrc-clk-map: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Defines clock map used in driver + <0> - select the map for asrc0 in imx8qm/imx8qxp + <1> - select the map for asrc1 in imx8qm/imx8qxp + enum: [0, 1] + + big-endian: + type: boolean + description: + 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: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + 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). + enum: [2, 6] + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + - fsl,asrc-rate + - fsl,asrc-width + +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + then: + required: + - fsl,asrc-clk-map + else: + properties: + fsl,asrc-clk-map: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/clock/imx6qdl-clock.h> + asrc: asrc@2034000 { + compatible = "fsl,imx53-asrc"; + reg = <0x02034000 0x4000>; + interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX6QDL_CLK_ASRC_IPG>, + <&clks IMX6QDL_CLK_ASRC_MEM>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks IMX6QDL_CLK_ASRC>, <&clks 0>, <&clks 0>, + <&clks IMX6QDL_CLK_SPBA>; + clock-names = "mem", "ipg", "asrck_0", + "asrck_1", "asrck_2", "asrck_3", "asrck_4", + "asrck_5", "asrck_6", "asrck_7", "asrck_8", + "asrck_9", "asrck_a", "asrck_b", "asrck_c", + "asrck_d", "asrck_e", "asrck_f", "spba"; + dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, + <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; + dma-names = "rxa", "rxb", "rxc", + "txa", "txb", "txc"; + fsl,asrc-rate = <48000>; + fsl,asrc-width = <16>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.txt b/Documentation/devicetree/bindings/sound/fsl,micfil.txt deleted file mode 100644 index 53e227b15277..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,micfil.txt +++ /dev/null @@ -1,32 +0,0 @@ -NXP MICFIL Digital Audio Interface (MICFIL). - -The MICFIL digital interface provides a 16-bit audio signal from a PDM -microphone bitstream in a configurable output sampling rate. - -Required properties: - - - compatible : Compatible list, contains "fsl,imx8mm-micfil" - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the micfil interrupts. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "ipg_clk" for register access and - "ipg_clk_app" for internal micfil clock. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - -Example: -micfil: micfil@30080000 { - compatible = "fsl,imx8mm-micfil"; - reg = <0x0 0x30080000 0x0 0x10000>; - interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, - <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&clk IMX8MM_CLK_PDM_IPG>, - <&clk IMX8MM_CLK_PDM_ROOT>; - clock-names = "ipg_clk", "ipg_clk_app"; - dmas = <&sdma2 24 26 0x80000000>; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml new file mode 100644 index 000000000000..c1e9803fc113 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml @@ -0,0 +1,95 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,micfil.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP MICFIL Digital Audio Interface (MICFIL) + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + The MICFIL digital interface provides a 16-bit or 24-bit audio signal + from a PDM microphone bitstream in a configurable output sampling rate. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx95-micfil + - const: fsl,imx93-micfil + + - enum: + - fsl,imx8mm-micfil + - fsl,imx8mp-micfil + - fsl,imx93-micfil + + reg: + maxItems: 1 + + interrupts: + items: + - description: Digital Microphone interface interrupt + - description: Digital Microphone interface error interrupt + - description: voice activity detector event interrupt + - description: voice activity detector error interrupt + + dmas: + items: + - description: DMA controller phandle and request line for RX + + dma-names: + items: + - const: rx + + clocks: + items: + - description: The ipg clock for register access + - description: internal micfil clock + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + - description: External clock 3 + minItems: 2 + + clock-names: + items: + - const: ipg_clk + - const: ipg_clk_app + - const: pll8k + - const: pll11k + - const: clkext3 + minItems: 2 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + micfil: audio-controller@30080000 { + compatible = "fsl,imx8mm-micfil"; + reg = <0x30080000 0x10000>; + interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 44 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 45 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>; + clock-names = "ipg_clk", "ipg_clk_app"; + dmas = <&sdma2 24 25 0>; + dma-names = "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt deleted file mode 100644 index 40353fc30255..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,mqs.txt +++ /dev/null @@ -1,36 +0,0 @@ -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,mqs.yaml b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml new file mode 100644 index 000000000000..8b33353a80ca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,mqs.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Medium Quality Sound (MQS) + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Chancel Liu <chancel.liu@nxp.com> + +description: | + Medium quality sound (MQS) is used to generate medium quality audio + via a standard GPIO in the pinmux, allowing the user to connect + stereo speakers or headphones to a power amplifier without an + additional DAC chip. + +properties: + compatible: + enum: + - fsl,imx6sx-mqs + - fsl,imx8qm-mqs + - fsl,imx8qxp-mqs + - fsl,imx93-mqs + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + minItems: 1 + maxItems: 2 + + gpr: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to the General Purpose Register (GPR) node + + reg: + maxItems: 1 + + power-domains: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-mqs + - fsl,imx8qxp-mqs + then: + properties: + clocks: + items: + - description: Master clock + - description: Clock for register access + clock-names: + items: + - const: mclk + - const: core + required: + - reg + - power-domains + else: + properties: + clocks: + items: + - description: Master clock + clock-names: + items: + - const: mclk + required: + - gpr + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx6sx-clock.h> + mqs0: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + }; + + - | + #include <dt-bindings/firmware/imx/rsrc.h> + mqs1: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&mqs0_lpcg 0>, <&mqs0_lpcg 1>; + clock-names = "mclk", "core"; + power-domains = <&pd IMX_SC_R_MQS_0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml b/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml new file mode 100644 index 000000000000..b522ed7dcc51 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml @@ -0,0 +1,118 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,qmc-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: QMC audio + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The QMC audio is an ASoC component which uses QMC (QUICC Multichannel + Controller) channels to transfer the audio data. + It provides as many DAI as the number of QMC channel used. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: fsl,qmc-audio + + '#address-cells': + const: 1 + '#size-cells': + const: 0 + '#sound-dai-cells': + const: 1 + +patternProperties: + '^dai@([0-9]|[1-5][0-9]|6[0-3])$': + description: + A DAI managed by this controller + type: object + additionalProperties: false + + properties: + reg: + minimum: 0 + maximum: 63 + description: + The DAI number + + fsl,qmc-chan: + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: phandle to QMC node + - description: Channel number + description: + Should be a phandle/number pair. The phandle to QMC node and the QMC + channel to use for this DAI. + + required: + - reg + - fsl,qmc-chan + +required: + - compatible + - '#address-cells' + - '#size-cells' + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + audio_controller: audio-controller { + compatible = "fsl,qmc-audio"; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + dai@16 { + reg = <16>; + fsl,qmc-chan = <&qmc 16>; + }; + dai@17 { + reg = <17>; + fsl,qmc-chan = <&qmc 17>; + }; + }; + + sound { + compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; + simple-audio-card,dai-link@0 { + reg = <0>; + format = "dsp_b"; + cpu { + sound-dai = <&audio_controller 16>; + }; + codec { + sound-dai = <&codec1>; + dai-tdm-slot-num = <4>; + dai-tdm-slot-width = <8>; + /* TS 3, 5, 7, 9 */ + dai-tdm-slot-tx-mask = <0 0 0 1 0 1 0 1 0 1>; + dai-tdm-slot-rx-mask = <0 0 0 1 0 1 0 1 0 1>; + }; + }; + simple-audio-card,dai-link@1 { + reg = <1>; + format = "dsp_b"; + cpu { + sound-dai = <&audio_controller 17>; + }; + codec { + sound-dai = <&codec2>; + dai-tdm-slot-num = <4>; + dai-tdm-slot-width = <8>; + /* TS 2, 4, 6, 8 */ + dai-tdm-slot-tx-mask = <0 0 1 0 1 0 1 0 1>; + dai-tdm-slot-rx-mask = <0 0 1 0 1 0 1 0 1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml new file mode 100644 index 000000000000..188f38baddec --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -0,0 +1,133 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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, MICFIL, DMA controlled by Cortex M core. What we see from + Linux side is a device which provides audio service by rpmsg channel. + We can create different sound cards which access different hardwares + such as SAI, MICFIL, .etc through building rpmsg channels between + Cortex-A and Cortex-M. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - fsl,imx7ulp-rpmsg-audio + - fsl,imx8mn-rpmsg-audio + - fsl,imx8mm-rpmsg-audio + - fsl,imx8mp-rpmsg-audio + - fsl,imx8ulp-rpmsg-audio + - fsl,imx93-rpmsg-audio + + 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 + + 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. + + fsl,rpmsg-channel-name: + $ref: /schemas/types.yaml#/definitions/string + description: | + A string property to assign rpmsg channel this sound card sits on. + This property can be omitted if there is only one sound card and it sits + on "rpmsg-audio-channel". + enum: + - rpmsg-audio-channel + - rpmsg-micfil-channel + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + rpmsg_audio: rpmsg_audio { + compatible = "fsl,imx8mn-rpmsg-audio"; + model = "wm8524-audio"; + fsl,enable-lpa; + fsl,rpmsg-out; + clocks = <&clk IMX8MN_CLK_SAI3_IPG>, + <&clk IMX8MN_CLK_SAI3_ROOT>, + <&clk IMX8MN_CLK_SDMA3_ROOT>, + <&clk IMX8MN_AUDIO_PLL1_OUT>, + <&clk IMX8MN_AUDIO_PLL2_OUT>; + clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; + }; + + - | + #include <dt-bindings/clock/imx8mm-clock.h> + + rpmsg_micfil: audio-controller { + compatible = "fsl,imx8mm-rpmsg-audio"; + model = "micfil-audio"; + fsl,rpmsg-channel-name = "rpmsg-micfil-channel"; + fsl,enable-lpa; + fsl,rpmsg-in; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>, + <&clk IMX8MM_CLK_SDMA3_ROOT>, + <&clk IMX8MM_AUDIO_PLL1_OUT>, + <&clk IMX8MM_AUDIO_PLL2_OUT>; + clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..2456d958adee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,209 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx6ul-sai + - fsl,imx7d-sai + - const: fsl,imx6sx-sai + + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + - items: + - enum: + - fsl,imx6sx-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - fsl,imx93-sai + - fsl,imx95-sai + - fsl,vf610-sai + + reg: + maxItems: 1 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 5 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + power-domains: + maxItems: 1 + + dmas: + minItems: 1 + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + minItems: 1 + items: + - const: rx + - const: tx + + interrupts: + items: + - description: receive and transmit interrupt + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + maxItems: 16 + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,sai-synchronous-rx: + description: | + SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + type: boolean + + fsl,sai-asynchronous: + description: | + SAI will work in the asynchronous mode, which means both transmitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + type: boolean + + fsl,shared-interrupt: + description: Interrupt is shared with other modules. + type: boolean + + lsb-first: + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +allOf: + - $ref: dai-common.yaml# + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +required: + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&edma0 0 20>, <&edma0 0 21>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + #sound-dai-cells = <0>; + }; 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..1d64e8337aa4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.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/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. + - description: PLL clock source for 8kHz series rate, optional. + - description: PLL clock source for 11khz series rate, optional. + 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 + - const: pll8k + - const: pll11k + minItems: 9 + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml new file mode 100644 index 000000000000..0eb0c1ba8710 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml @@ -0,0 +1,125 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + - fsl,imx93-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: + items: + - description: WAKEUPMIX Audio XCVR Interrupt 1 + - description: WAKEUPMIX Audio XCVR Interrupt 2 + minItems: 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 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx93-xcvr + then: + properties: + interrupts: + minItems: 2 + maxItems: 2 + else: + properties: + interrupts: + maxItems: 1 + +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..4e8dbc5abfd1 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -34,6 +34,20 @@ 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" + + "fsl,imx-audio-nau8822" + Required properties: - compatible : Contains one of entries in the compatible list. @@ -44,6 +58,11 @@ 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 +79,14 @@ 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 deleted file mode 100644 index 0dc83cc4a236..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,82 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai" or - "fsl,imx8qm-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - - 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". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; diff --git a/Documentation/devicetree/bindings/sound/google,chv3-codec.yaml b/Documentation/devicetree/bindings/sound/google,chv3-codec.yaml new file mode 100644 index 000000000000..5329dc140b1c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,chv3-codec.yaml @@ -0,0 +1,31 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,chv3-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google Chameleon v3 audio codec + +maintainers: + - PaweÅ‚ Anikiel <pan@semihalf.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: google,chv3-codec + + "#sound-dai-cells": + const: 0 + +required: + - compatible + +additionalProperties: false + +examples: + - | + audio-codec { + compatible = "google,chv3-codec"; + }; diff --git a/Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml b/Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml new file mode 100644 index 000000000000..3ce910f44d39 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,chv3-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google Chameleon v3 I2S device + +maintainers: + - PaweÅ‚ Anikiel <pan@semihalf.com> + +description: | + I2S device for the Google Chameleon v3. The device handles both RX + and TX using a producer/consumer ring buffer design. + +properties: + compatible: + const: google,chv3-i2s + + reg: + items: + - description: core registers + - description: irq registers + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@c0060300 { + compatible = "google,chv3-i2s"; + reg = <0xc0060300 0x100>, + <0xc0060f00 0x10>; + interrupts = <GIC_SPI 20 IRQ_TYPE_LEVEL_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c84e656afb0a..1434f4433738 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,12 +8,17 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang@chromium.org> + - Tzung-Bi Shih <tzungbi@kernel.org> 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 as a sub-node of the EC node (see - Documentation/devicetree/bindings/mfd/cros-ec.txt). + 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). + +allOf: + - $ref: dai-common.yaml# properties: compatible: @@ -31,7 +36,7 @@ properties: The last one integer is the length of the shared memory. memory-region: - $ref: '/schemas/types.yaml#/definitions/phandle' + maxItems: 1 description: | Shared memory region to EC. A "shared-dma-pool". See ../reserved-memory/reserved-memory.txt for details. @@ -40,7 +45,7 @@ required: - compatible - '#sound-dai-cells' -additionalProperties: false +unevaluatedProperties: false examples: - | @@ -54,14 +59,20 @@ examples: #size-cells = <0>; cros-ec@0 { compatible = "google,cros-ec-spi"; - #address-cells = <2>; - #size-cells = <1>; reg = <0>; - cros_ec_codec: ec-codec@10500000 { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; + interrupts = <93 0>; + + codecs { + #address-cells = <2>; + #size-cells = <1>; + + cros_ec_codec: ec-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + }; }; }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml new file mode 100644 index 000000000000..bac940553965 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7180-trogdor.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7180-Trogdor ASoC sound card driver + +maintainers: + - Rohit kumar <quic_rohkumar@quicinc.com> + - Cheng-Yi Chiang <cychiang@chromium.org> + +description: + This binding describes the SC7180 sound card which uses LPASS for audio. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - google,sc7180-trogdor + - google,sc7180-coachz + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + dmic-gpios: + maxItems: 1 + description: GPIO for switching between DMICs + +patternProperties: + "^dai-link(@[0-9])?$": + 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. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 4 + + required: + - link-name + - cpu + - codec + + additionalProperties: false + +required: + - compatible + - "#address-cells" + - "#size-cells" + +unevaluatedProperties: false + +examples: + + - | + sound { + compatible = "google,sc7180-trogdor"; + model = "sc7180-rt5682-max98357a-2mic"; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR"; + + #address-cells = <1>; + #size-cells = <0>; + + dmic-gpios = <&tlmm 86 0>; + + dai-link@0 { + link-name = "MultiMedia0"; + reg = <0>; + cpu { + sound-dai = <&lpass_cpu 0>; + }; + + codec { + sound-dai = <&alc5682 0>; + }; + }; + + dai-link@1 { + link-name = "MultiMedia1"; + reg = <1>; + cpu { + sound-dai = <&lpass_cpu 1>; + }; + + codec { + sound-dai = <&max98357a>; + }; + }; + + dai-link@2 { + link-name = "MultiMedia2"; + reg = <2>; + cpu { + sound-dai = <&lpass_hdmi 0>; + }; + + codec { + sound-dai = <&msm_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml new file mode 100644 index 000000000000..cdcd7c6f21eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7280-herobrine.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7280-Herobrine ASoC sound card driver + +maintainers: + - Judy Hsiao <judyhsiao@chromium.org> + +description: + This binding describes the SC7280 sound card which uses LPASS for audio. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - google,sc7280-herobrine + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]$": + 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 + + platform: + description: Holds subnode which includes the phandle of q6apm platform device. + type: object + properties: + sound-dai: + maxItems: 1 + + required: + - sound-dai + + additionalProperties: false + + required: + - link-name + - cpu + - codec + - reg + + additionalProperties: false + +required: + - compatible + - "#address-cells" + - "#size-cells" + +unevaluatedProperties: 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 deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -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 : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; 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..f7ad5ea2491e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -0,0 +1,117 @@ +# 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> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - fsl,imx-audio-card + +patternProperties: + ".*-dai-link$": + 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 + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + + fsl,mclk-equal-bclk: + description: Indicates mclk can be equal to bclk, especially for sai interface + $ref: /schemas/types.yaml#/definitions/flag + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + sound-ak4458 { + compatible = "fsl,imx-audio-card"; + model = "ak4458-audio"; + pri-dai-link { + link-name = "akcodec"; + format = "i2s"; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + fe-dai-link { + link-name = "HiFi-ASRC-FE"; + format = "i2s"; + cpu { + sound-dai = <&easrc>; + }; + }; + be-dai-link { + link-name = "HiFi-ASRC-BE"; + format = "dsp_b"; + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000000..e7e7bb65c366 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml b/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml new file mode 100644 index 000000000000..5e11ce2c13ac --- /dev/null +++ b/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/infineon,peb2466.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Infineon PEB2466 codec + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The Infineon PEB2466 codec is a programmable DSP-based four channels codec + with filters capabilities. + + The time-slots used by the codec must be set and so, the properties + 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and + 'dai-tdm-slot-rx-mask' must be present in the sound card node for sub-nodes + that involve the codec. The codec uses one 8bit time-slot per channel. + 'dai-tdm-tdm-slot-with' must be set to 8. + + The PEB2466 codec also supports 28 gpios (signaling pins). + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml + - $ref: dai-common.yaml# + +properties: + compatible: + const: infineon,peb2466 + + reg: + description: + SPI device address. + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + spi-max-frequency: + maximum: 8192000 + + reset-gpios: + description: + GPIO used to reset the device. + maxItems: 1 + + firmware-name: + maxItems: 1 + description: + Filters coefficients file to load. If this property is omitted, internal + filters are disabled. + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - '#sound-dai-cells' + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + compatible = "infineon,peb2466"; + reg = <0>; + spi-max-frequency = <8192000>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,aic.yaml b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml index 44f49bebb267..d15c000f14e1 100644 --- a/Documentation/devicetree/bindings/sound/ingenic,aic.yaml +++ b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml @@ -4,11 +4,14 @@ $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 +title: Ingenic SoCs AC97 / I2S Controller (AIC) maintainers: - Paul Cercueil <paul@crapouillou.net> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: '^audio-controller@' @@ -16,13 +19,14 @@ properties: compatible: oneOf: - enum: - - ingenic,jz4740-i2s - - ingenic,jz4760-i2s - - ingenic,jz4770-i2s - - ingenic,jz4780-i2s + - ingenic,jz4740-i2s + - ingenic,jz4760-i2s + - ingenic,jz4770-i2s + - ingenic,jz4780-i2s + - ingenic,x1000-i2s - items: - - const: ingenic,jz4725b-i2s - - const: ingenic,jz4740-i2s + - const: ingenic,jz4725b-i2s + - const: ingenic,jz4740-i2s '#sound-dai-cells': const: 0 @@ -37,15 +41,11 @@ properties: 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: @@ -57,7 +57,7 @@ properties: - const: rx - const: tx -additionalProperties: false +unevaluatedProperties: false required: - compatible @@ -71,7 +71,7 @@ required: examples: - | - #include <dt-bindings/clock/jz4740-cgu.h> + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> aic: audio-controller@10020000 { compatible = "ingenic,jz4740-i2s"; reg = <0x10020000 0x38>; @@ -82,10 +82,8 @@ examples: 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"; + <&cgu JZ4740_CLK_I2S>; + clock-names = "aic", "i2s"; 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 index eb4be86464bb..b58b90850e35 100644 --- a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml +++ b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml @@ -4,20 +4,28 @@ $id: http://devicetree.org/schemas/sound/ingenic,codec.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Ingenic JZ47xx internal codec DT bindings +title: Ingenic JZ47xx internal codec maintainers: - Paul Cercueil <paul@crapouillou.net> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: '^audio-codec@.*' compatible: oneOf: - - const: ingenic,jz4770-codec - - const: ingenic,jz4725b-codec - - const: ingenic,jz4740-codec + - enum: + - ingenic,jz4770-codec + - ingenic,jz4760-codec + - ingenic,jz4725b-codec + - ingenic,jz4740-codec + - items: + - const: ingenic,jz4760b-codec + - const: ingenic,jz4760-codec reg: maxItems: 1 @@ -32,7 +40,7 @@ properties: '#sound-dai-cells': const: 0 -additionalProperties: false +unevaluatedProperties: false required: - compatible @@ -43,7 +51,7 @@ required: examples: - | - #include <dt-bindings/clock/jz4740-cgu.h> + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> codec: audio-codec@10020080 { compatible = "ingenic,jz4740-codec"; reg = <0x10020080 0x8>; diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml new file mode 100644 index 000000000000..76b6f2cf25df --- /dev/null +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright 2020 Intel Corporation +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Intel KeemBay I2S + +maintainers: + - Daniele Alessandrelli <daniele.alessandrelli@intel.com> + - Paul J. Murphy <paul.j.murphy@intel.com> + +description: | + Intel KeemBay I2S + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - intel,keembay-i2s + - intel,keembay-tdm + - intel,keembay-hdmi-i2s + + "#sound-dai-cells": + const: 0 + + reg: + items: + - description: I2S registers + - description: I2S gen configuration + + reg-names: + items: + - const: i2s-regs + - const: i2s_gen_cfg + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: osc + - const: apb_clk + + dmas: + items: + - description: DMA TX channel + - description: DMA RX channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #define KEEM_BAY_PSS_AUX_I2S3 + #define KEEM_BAY_PSS_I2S3 + i2s3: i2s@20140000 { + compatible = "intel,keembay-i2s"; + #sound-dai-cells = <0>; + reg = <0x20140000 0x200>, /* I2S registers */ + <0x202a00a4 0x4>; /* I2S gen configuration */ + reg-names = "i2s-regs", "i2s_gen_cfg"; + interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; + clock-names = "osc", "apb_clk"; + clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>; + dmas = <&axi_dma0 29>, <&axi_dma0 33>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml new file mode 100644 index 000000000000..b36c35e5da1a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1303.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1303 Audio Amplifier + +maintainers: + - Kiseok Jo <kiseok.jo@irondevice.com> + +description: + SMA1303 digital class-D audio amplifier + with an integrated boost converter. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1303 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1303"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml new file mode 100644 index 000000000000..a67b79cbe006 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bluetooth SCO Audio Codec + +maintainers: + - Mark Brown <broonie@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + '#sound-dai-cells': + enum: + - 0 + + # For Wideband PCM + - 1 + + compatible: + enum: + - delta,dfbmcs320 + - linux,bt-sco + +required: + - '#sound-dai-cells' + - compatible + +unevaluatedProperties: false + +examples: + - | + codec { + #sound-dai-cells = <0>; + compatible = "linux,bt-sco"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml new file mode 100644 index 000000000000..fe5f0756af2f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,spdif-dit.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dummy SPDIF Transmitter + +maintainers: + - Mark Brown <broonie@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: linux,spdif-dit + + "#sound-dai-cells": + const: 0 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + +additionalProperties: false + +examples: + - | + spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000000..61e8babed402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson 7axxx/2kxxx ASoC audio sound card driver + +maintainers: + - Yingkun Meng <mengyingkun@loongson.cn> + +description: + The binding describes the sound card present in loongson + 7axxx/2kxxx platform. The sound card is an ASoC component + which uses Loongson I2S controller to transfer the audio data. + +properties: + compatible: + const: loongson,ls-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - model + - mclk-fs + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml index 6d20a24a2ae9..4193d17d1c62 100644 --- a/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml +++ b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml @@ -4,11 +4,14 @@ $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 +title: Marvel SSPA Digital Audio Interface maintainers: - Lubomir Rintel <lkundrak@v3.sk> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: "^audio-controller(@.*)?$" @@ -51,36 +54,18 @@ properties: - const: rx port: - type: object + $ref: audio-graph-port.yaml# + unevaluatedProperties: false properties: endpoint: type: object + additionalProperties: true properties: - remote-endpoint: true - - frame-master: - type: boolean - description: SoC generates the frame clock - - bitclock-master: - type: boolean - description: SoC generates the bit clock - dai-format: - $ref: /schemas/types.yaml#/definitions/string - description: The digital audio format const: i2s - required: - - remote-endpoint - - required: - - endpoint - - additionalProperties: false - required: - "#sound-dai-cells" - compatible @@ -92,7 +77,7 @@ required: - dma-names - port -additionalProperties: false +unevaluatedProperties: false examples: - | @@ -112,8 +97,6 @@ examples: port { endpoint { remote-endpoint = <&rt5631_0>; - frame-master; - bitclock-master; dai-format = "i2s"; }; }; diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt deleted file mode 100644 index 7e1bbd5c27fd..000000000000 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ /dev/null @@ -1,59 +0,0 @@ -MAX98090 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "maxim,max98090" or "maxim,max98091". - -- reg : The I2C address of the device. - -- interrupts : The CODEC's interrupt output. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC - -- clock-names: Should be "mclk" - -- #sound-dai-cells : should be 0. - -- maxim,dmic-freq: Frequency at which to clock DMIC - -- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: - 0 - 2.2v - 1 - 2.55v - 2 - 2.4v - 3 - 2.8v - -Pins on the device (for linking into audio routes): - - * MIC1 - * MIC2 - * DMICL - * DMICR - * IN1 - * IN2 - * IN3 - * IN4 - * IN5 - * IN6 - * IN12 - * IN34 - * IN56 - * HPL - * HPR - * SPKL - * SPKR - * RCVL - * RCVR - * MICBIAS - -Example: - -audio-codec@10 { - compatible = "maxim,max98090"; - reg = <0x10>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt deleted file mode 100644 index 318a4c82f17f..000000000000 --- a/Documentation/devicetree/bindings/sound/max98095.txt +++ /dev/null @@ -1,22 +0,0 @@ -MAX98095 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "maxim,max98095". - -- reg : The I2C address of the device. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC - -- clock-names: Should be "mclk" - -Example: - -max98095: codec@11 { - compatible = "maxim,max98095"; - reg = <0x11>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt deleted file mode 100644 index 4bce14ce806f..000000000000 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ /dev/null @@ -1,22 +0,0 @@ -Maxim MAX98357A audio DAC - -This node models the Maxim MAX98357A DAC. - -Required properties: -- compatible : "maxim,max98357a" - -Optional properties: -- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. - If this option is not specified then driver does not manage - the pin state (e.g. chip is always on). -- sdmode-delay : specify delay time for SD_MODE pin. - If this option is specified, which means it's required i2s clocks - ready before SD_MODE is unmuted in order to avoid the speaker pop noise. - It's observed that 5ms is sufficient. - -Example: - -max98357a { - compatible = "maxim,max98357a"; - sdmode-gpios = <&qcom_pinmux 25 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt deleted file mode 100644 index 8b2b2704b574..000000000000 --- a/Documentation/devicetree/bindings/sound/max98371.txt +++ /dev/null @@ -1,17 +0,0 @@ -max98371 codec - -This device supports I2C mode only. - -Required properties: - -- compatible : "maxim,max98371" -- reg : The chip select number on the I2C bus - -Example: - -&i2c { - max98371: max98371@31 { - compatible = "maxim,max98371"; - reg = <0x31>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt deleted file mode 100644 index 583ed5fdfb28..000000000000 --- a/Documentation/devicetree/bindings/sound/max98504.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim MAX98504 class D mono speaker amplifier - -This device supports I2C control interface and an IRQ output signal. It features -a PCM and PDM digital audio interface (DAI) and a differential analog input. - -Required properties: - - - compatible : "maxim,max98504" - - reg : should contain the I2C slave device address - - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, - as covered in ../regulator/regulator.txt - - interrupts : should specify the interrupt line the device is connected to, - as described in ../interrupt-controller/interrupts.txt - -Optional properties: - - - maxim,brownout-threshold - the PVDD brownout threshold, the value must be - from 0, 1...21 range, corresponding to 2.6V, 2.65V...3.65V voltage range - - maxim,brownout-attenuation - the brownout attenuation to the speaker gain - applied during the "attack hold" and "timed hold" phase, the value must be - from 0...6 (dB) range - - maxim,brownout-attack-hold-ms - the brownout attack hold phase time in ms, - 0...255 (VBATBROWN_ATTK_HOLD, register 0x0018) - - maxim,brownout-timed-hold-ms - the brownout timed hold phase time in ms, - 0...255 (VBATBROWN_TIME_HOLD, register 0x0019) - - maxim,brownout-release-rate-ms - the brownout release phase step time in ms, - 0...255 (VBATBROWN_RELEASE, register 0x001A) - -The default value when the above properties are not specified is 0, -the maxim,brownout-threshold property must be specified to actually enable -the PVDD brownout protection. - -Example: - - max98504@31 { - compatible = "maxim,max98504"; - reg = <0x31>; - interrupt-parent = <&gpio_bank_0>; - interrupts = <2 0>; - - DVDD-supply = <®ulator>; - DIOVDD-supply = <®ulator>; - PVDD-supply = <®ulator>; -}; diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt deleted file mode 100644 index b8bd914ee697..000000000000 --- a/Documentation/devicetree/bindings/sound/max9867.txt +++ /dev/null @@ -1,17 +0,0 @@ -max9867 codec - -This device supports I2C mode only. - -Required properties: - -- compatible : "maxim,max9867" -- reg : The chip select number on the I2C bus - -Example: - -&i2c { - max9867: max9867@18 { - compatible = "maxim,max9867"; - reg = <0x18>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index f6171591ddc6..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,41 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max9759.txt b/Documentation/devicetree/bindings/sound/maxim,max9759.txt deleted file mode 100644 index 737a996374d3..000000000000 --- a/Documentation/devicetree/bindings/sound/maxim,max9759.txt +++ /dev/null @@ -1,18 +0,0 @@ -Maxim MAX9759 Speaker Amplifier -=============================== - -Required properties: -- compatible : "maxim,max9759" -- shutdown-gpios : the gpio connected to the shutdown pin -- mute-gpios : the gpio connected to the mute pin -- gain-gpios : the 2 gpios connected to the g1 and g2 pins - -Example: - -max9759: analog-amplifier { - compatible = "maxim,max9759"; - shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>; - gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>, - <&gpio3 25 GPIO_ACTIVE_LOW>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max9759.yaml b/Documentation/devicetree/bindings/sound/maxim,max9759.yaml new file mode 100644 index 000000000000..a76ee6a635af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9759.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/maxim,max9759.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim MAX9759 Speaker Amplifier + +maintainers: + - Otabek Nazrullaev <otabeknazrullaev1998@gmail.com> + +properties: + compatible: + const: maxim,max9759 + + shutdown-gpios: + maxItems: 1 + description: the gpio connected to the shutdown pin + + mute-gpios: + maxItems: 1 + description: the gpio connected to the mute pin + + gain-gpios: + maxItems: 2 + description: the 2 gpios connected to the g1 and g2 pins + +required: + - compatible + - shutdown-gpios + - mute-gpios + - gain-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + amplifier { + compatible = "maxim,max9759"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>; + gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>, + <&gpio3 25 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98090.yaml b/Documentation/devicetree/bindings/sound/maxim,max98090.yaml new file mode 100644 index 000000000000..65e4c516912f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98090.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98090/MAX98091 audio codecs + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +description: | + Pins on the device (for linking into audio routes): + MIC1, MIC2, DMICL, DMICR, IN1, IN2, IN3, IN4, IN5, IN6, IN12, IN34, IN56, + HPL, HPR, SPKL, SPKR, RCVL, RCVR, MICBIAS + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98090 + - maxim,max98091 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + interrupts: + maxItems: 1 + + maxim,dmic-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 2500000 + description: + DMIC clock frequency + + maxim,micbias: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1, 2, 3 ] + default: 3 + description: | + Micbias voltage applied to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 IRQ_TYPE_EDGE_FALLING>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98095.yaml b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml new file mode 100644 index 000000000000..77544a9e1587 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98095.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98095 audio codec + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98095 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98357a.yaml b/Documentation/devicetree/bindings/sound/maxim,max98357a.yaml new file mode 100644 index 000000000000..83ba8666fbb4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98357a.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98357a.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98357A/MAX98360A amplifier + +maintainers: + - Tzung-Bi Shih <tzungbi@kernel.org> + +description: + Maxim Integrated MAX98357A/MAX98360A is a digital pulse-code modulation (PCM) + input Class D amplifier. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98357a + - maxim,max98360a + + '#sound-dai-cells': + const: 0 + + sdmode-gpios: + maxItems: 1 + description: + Chip's SD_MODE pin. If missing the chip is always on. + + sdmode-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Delay time for SD_MODE pin changes intended to make I2S clocks ready + before SD_MODE is unmuted in order to avoid the speaker pop noise. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + amplifier { + compatible = "maxim,max98360a"; + #sound-dai-cells = <0>; + sdmode-gpios = <&qcom_pinmux 25 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98371.yaml b/Documentation/devicetree/bindings/sound/maxim,max98371.yaml new file mode 100644 index 000000000000..14fba34ef81a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98371.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98371.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim MAX98371 audio codec + +maintainers: + - anish kumar <yesanishhere@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: maxim,max98371 + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@31 { + compatible = "maxim,max98371"; + reg = <0x31>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000000..deaa6886c42f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 8388607 + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + reset-gpios = <&gpio 9 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98504.yaml b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml new file mode 100644 index 000000000000..23f19a9d2c06 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98504.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98504 class D mono speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +description: + Maxim Integrated MAX98504 speaker amplifier supports I2C control interface + with an IRQ output signal, PCM and PDM digital audio interface (DAI) and a + differential analog input. + +properties: + compatible: + const: maxim,max98504 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + DIOVDD-supply: true + DVDD-supply: true + PVDD-supply: true + + maxim,brownout-threshold: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 21 + default: 0 + description: + PVDD brownout threshold, where values correspond to 2.6V, 2.65V...3.65V + voltage range. Property also enables the PVDD brownout protection. + + maxim,brownout-attenuation: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 6 + default: 0 + description: + Brownout attenuation to the speaker gain applied during the "attack hold" + and "timed hold" phase, the value must be from 0...6 (dB) range. + + maxim,brownout-attack-hold-ms: + maximum: 255 + default: 0 + description: + Brownout attack hold phase time in ms, VBATBROWN_ATTK_HOLD, register 0x0018. + + maxim,brownout-timed-hold-ms: + maximum: 255 + default: 0 + description: + Brownout timed hold phase time in ms, VBATBROWN_TIME_HOLD, register 0x0019. + + maxim,brownout-release-rate-ms: + maximum: 255 + default: 0 + description: + Brownout release phase step time in ms, VBATBROWN_RELEASE, register 0x001A. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DIOVDD-supply = <&ldo3_reg>; + DVDD-supply = <&ldo3_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..3f88c7d61e34 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song@maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max9867.yaml b/Documentation/devicetree/bindings/sound/maxim,max9867.yaml new file mode 100644 index 000000000000..0b9a84d33b6c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9867.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max9867.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX9867 CODEC + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * LOUT + * ROUT + * LINL + * LINR + * MICL + * MICR + * DMICL + * DMICR + +maintainers: + - Ladislav Michl <ladis@linux-mips.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max9867 + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@18 { + compatible = "maxim,max9867"; + #sound-dai-cells = <0>; + reg = <0x18>; + clocks = <&codec_clk>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..32fd86204a7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee@maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; 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/mediatek,mt2701-audio.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml new file mode 100644 index 000000000000..45382c4d86aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.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/mediatek,mt2701-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End (AFE) PCM controller for mt2701 + +description: + The AFE PCM node must be a subnode of the MediaTek audsys device tree node. + +maintainers: + - Eugen Hristev <eugen.hristev@collabora.com> + +properties: + compatible: + enum: + - mediatek,mt2701-audio + - mediatek,mt7622-audio + + interrupts: + items: + - description: AFE interrupt + - description: ASYS interrupt + + interrupt-names: + items: + - const: afe + - const: asys + + power-domains: + maxItems: 1 + + clocks: + items: + - description: audio infra sys clock + - description: top audio mux 1 + - description: top audio mux 2 + - description: top audio sys a1 clock + - description: top audio sys a2 clock + - description: i2s0 source selection + - description: i2s1 source selection + - description: i2s2 source selection + - description: i2s3 source selection + - description: i2s0 source divider + - description: i2s1 source divider + - description: i2s2 source divider + - description: i2s3 source divider + - description: i2s0 master clock + - description: i2s1 master clock + - description: i2s2 master clock + - description: i2s3 master clock + - description: i2so0 hopping clock + - description: i2so1 hopping clock + - description: i2so2 hopping clock + - description: i2so3 hopping clock + - description: i2si0 hopping clock + - description: i2si1 hopping clock + - description: i2si2 hopping clock + - description: i2si3 hopping clock + - description: asrc0 output clock + - description: asrc1 output clock + - description: asrc2 output clock + - description: asrc3 output clock + - description: audio front end pd clock + - description: audio front end conn pd clock + - description: top audio a1 sys pd + - description: top audio a2 sys pd + - description: audio merge interface pd + + clock-names: + items: + - const: infra_sys_audio_clk + - const: top_audio_mux1_sel + - const: top_audio_mux2_sel + - const: top_audio_a1sys_hp + - const: top_audio_a2sys_hp + - const: i2s0_src_sel + - const: i2s1_src_sel + - const: i2s2_src_sel + - const: i2s3_src_sel + - const: i2s0_src_div + - const: i2s1_src_div + - const: i2s2_src_div + - const: i2s3_src_div + - const: i2s0_mclk_en + - const: i2s1_mclk_en + - const: i2s2_mclk_en + - const: i2s3_mclk_en + - const: i2so0_hop_ck + - const: i2so1_hop_ck + - const: i2so2_hop_ck + - const: i2so3_hop_ck + - const: i2si0_hop_ck + - const: i2si1_hop_ck + - const: i2si2_hop_ck + - const: i2si3_hop_ck + - const: asrc0_out_ck + - const: asrc1_out_ck + - const: asrc2_out_ck + - const: asrc3_out_ck + - const: audio_afe_pd + - const: audio_afe_conn_pd + - const: audio_a1sys_pd + - const: audio_a2sys_pd + - const: audio_mrgif_pd + +required: + - compatible + - interrupts + - interrupt-names + - power-domains + - clocks + - clock-names + +additionalProperties: false diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml new file mode 100644 index 000000000000..398efdfe00f5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml @@ -0,0 +1,160 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt7986-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AFE PCM controller for MT7986 + +maintainers: + - Maso Huang <maso.huang@mediatek.com> + +properties: + compatible: + oneOf: + - const: mediatek,mt7986-afe + - items: + - enum: + - mediatek,mt7981-afe + - mediatek,mt7988-afe + - const: mediatek,mt7986-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 5 + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + - description: audio i2s/pcm mck + + clock-names: + minItems: 5 + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + - const: i2s_m + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + const: mediatek,mt7986-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + + - if: + properties: + compatible: + contains: + const: mediatek,mt7981-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + + - if: + properties: + compatible: + contains: + const: mediatek,mt7988-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + - description: audio i2s/pcm mck + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + - const: i2s_m + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/clock/mt7986-clk.h> + + afe@11210000 { + compatible = "mediatek,mt7986-afe"; + reg = <0x11210000 0x9000>; + interrupts = <GIC_SPI 106 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&infracfg_ao CLK_INFRA_AUD_BUS_CK>, + <&infracfg_ao CLK_INFRA_AUD_26M_CK>, + <&infracfg_ao CLK_INFRA_AUD_L_CK>, + <&infracfg_ao CLK_INFRA_AUD_AUD_CK>, + <&infracfg_ao CLK_INFRA_AUD_EG2_CK>; + clock-names = "bus_ck", + "26m_ck", + "l_ck", + "aud_ck", + "eg2_ck"; + assigned-clocks = <&topckgen CLK_TOP_A1SYS_SEL>, + <&topckgen CLK_TOP_AUD_L_SEL>, + <&topckgen CLK_TOP_A_TUNER_SEL>; + assigned-clock-parents = <&topckgen CLK_TOP_APLL2_D4>, + <&apmixedsys CLK_APMIXED_APLL2>, + <&topckgen CLK_TOP_APLL2_D4>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml new file mode 100644 index 000000000000..09247ceea3f7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.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/mediatek,mt7986-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT7986 sound card with WM8960 codec + +maintainers: + - Maso Huang <maso.huang@mediatek.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: mediatek,mt7986-wm8960-sound + + platform: + type: object + additionalProperties: false + properties: + sound-dai: + description: The phandle of MT7986 platform. + maxItems: 1 + required: + - sound-dai + + codec: + type: object + additionalProperties: false + properties: + sound-dai: + description: The phandle of wm8960 codec. + maxItems: 1 + required: + - sound-dai + +unevaluatedProperties: false + +required: + - compatible + - audio-routing + - platform + - codec + +examples: + - | + sound { + compatible = "mediatek,mt7986-wm8960-sound"; + model = "mt7986-wm8960"; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + + platform { + sound-dai = <&afe>; + }; + + codec { + sound-dai = <&wm8960>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml new file mode 100644 index 000000000000..77af276ed2a2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml @@ -0,0 +1,241 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8188-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AFE PCM controller for mt8188 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8188-afe + + 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 dtschema reserved-memory/shared-dma-pool.yaml for details. + + mediatek,topckgen: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the mediatek topckgen controller + + mediatek,infracfg: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the mediatek infracfg 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 26m clock + - description: audio pll1 divide 4 + - description: audio pll2 divide 4 + - description: clock divider for iec + - description: mux for a2sys clock + - description: mux for aud_iec + + clock-names: + items: + - const: clk26m + - const: apll1 + - const: apll2 + - const: apll12_div0 + - const: apll12_div1 + - const: apll12_div2 + - const: apll12_div3 + - const: apll12_div9 + - const: top_a1sys_hp + - const: top_aud_intbus + - const: top_audio_h + - const: top_audio_local_bus + - const: top_dptx + - const: top_i2so1 + - const: top_i2so2 + - const: top_i2si1 + - const: top_i2si2 + - const: adsp_audio_26m + - const: apll1_d4 + - const: apll2_d4 + - const: apll12_div4 + - const: top_a2sys + - const: top_aud_iec + + mediatek,etdm-in1-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 module. + enum: + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + mediatek,etdm-in2-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 module. + enum: + - 0 # etdm1_in + - 2 # etdm1_out + - 3 # etdm2_out + + mediatek,etdm-out1-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 module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 3 # etdm2_out + + mediatek,etdm-out2-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + +patternProperties: + "^mediatek,etdm-in[1-2]-chn-disabled$": + $ref: /schemas/types.yaml#/definitions/uint8-array + minItems: 1 + maxItems: 16 + description: + This is a list of channel IDs which should be disabled. + By default, all data received from ETDM pins will be outputted to + memory. etdm in supports disable_out in direct mode(w/o interconn), + so user can disable the specified channels by the property. + uniqueItems: true + items: + minimum: 0 + maximum: 15 + + "^mediatek,etdm-in[1-2]-multi-pin-mode$": + 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. + +required: + - compatible + - reg + - interrupts + - resets + - reset-names + - mediatek,topckgen + - mediatek,infracfg + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe@10b10000 { + compatible = "mediatek,mt8188-afe"; + reg = <0x10b10000 0x10000>; + interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; + resets = <&watchdog 14>; + reset-names = "audiosys"; + memory-region = <&snd_dma_mem_reserved>; + mediatek,topckgen = <&topckgen>; + mediatek,infracfg = <&infracfg_ao>; + power-domains = <&spm 13>; //MT8188_POWER_DOMAIN_AUDIO + mediatek,etdm-in2-cowork-source = <2>; + mediatek,etdm-out2-cowork-source = <0>; + mediatek,etdm-in1-multi-pin-mode; + mediatek,etdm-in1-chn-disabled = /bits/ 8 <0x0 0x2>; + clocks = <&clk26m>, + <&apmixedsys 9>, //CLK_APMIXED_APLL1 + <&apmixedsys 10>, //CLK_APMIXED_APLL2 + <&topckgen 186>, //CLK_TOP_APLL12_CK_DIV0 + <&topckgen 187>, //CLK_TOP_APLL12_CK_DIV1 + <&topckgen 188>, //CLK_TOP_APLL12_CK_DIV2 + <&topckgen 189>, //CLK_TOP_APLL12_CK_DIV3 + <&topckgen 191>, //CLK_TOP_APLL12_CK_DIV9 + <&topckgen 83>, //CLK_TOP_A1SYS_HP + <&topckgen 31>, //CLK_TOP_AUD_INTBUS + <&topckgen 32>, //CLK_TOP_AUDIO_H + <&topckgen 69>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 81>, //CLK_TOP_DPTX + <&topckgen 77>, //CLK_TOP_I2SO1 + <&topckgen 78>, //CLK_TOP_I2SO2 + <&topckgen 79>, //CLK_TOP_I2SI1 + <&topckgen 80>, //CLK_TOP_I2SI2 + <&adsp_audio26m 0>, //CLK_AUDIODSP_AUDIO26M + <&topckgen 132>, //CLK_TOP_APLL1_D4 + <&topckgen 133>, //CLK_TOP_APLL2_D4 + <&topckgen 183>, //CLK_TOP_APLL12_CK_DIV4 + <&topckgen 84>, //CLK_TOP_A2SYS + <&topckgen 82>; //CLK_TOP_AUD_IEC>; + clock-names = "clk26m", + "apll1", + "apll2", + "apll12_div0", + "apll12_div1", + "apll12_div2", + "apll12_div3", + "apll12_div9", + "top_a1sys_hp", + "top_aud_intbus", + "top_audio_h", + "top_audio_local_bus", + "top_dptx", + "top_i2so1", + "top_i2so2", + "top_i2si1", + "top_i2si2", + "adsp_audio_26m", + "apll1_d4", + "apll2_d4", + "apll12_div4", + "top_a2sys", + "top_aud_iec"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml new file mode 100644 index 000000000000..f94ad0715e32 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.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/mediatek,mt8188-mt6359.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8188 ASoC sound card + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8188-es8326 + - mediatek,mt8188-mt6359-evb + - mediatek,mt8188-nau8825 + - mediatek,mt8188-rt5682s + + audio-routing: + description: + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8188 ASoC platform. + +patternProperties: + "^dai-link-[0-9]+$": + type: object + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: + This property corresponds to the name of the BE dai-link to which + we are going to update parameters in this node. + items: + enum: + - DPTX_BE + - ETDM1_IN_BE + - ETDM2_IN_BE + - ETDM1_OUT_BE + - ETDM2_OUT_BE + - ETDM3_OUT_BE + - PCM1_BE + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + items: + enum: + - cpu + - codec + + additionalProperties: false + + required: + - link-name + +unevaluatedProperties: false + +required: + - compatible + - mediatek,platform + +examples: + - | + sound { + compatible = "mediatek,mt8188-mt6359-evb"; + model = "MT6359-EVB"; + mediatek,platform = <&afe>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + audio-routing = + "Headphone", "Headphone L", + "Headphone", "Headphone R", + "AIN1", "Headset Mic"; + dai-link-0 { + link-name = "ETDM3_OUT_BE"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&hdmi0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml new file mode 100644 index 000000000000..fb630a184350 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-i2smcc.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: + oneOf: + - enum: + - microchip,sam9x60-i2smcc + - microchip,sama7g5-i2smcc + - items: + - enum: + - microchip,sam9x7-i2smcc + - const: microchip,sam9x60-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 + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + const: microchip,sam9x60-i2smcc + then: + properties: + microchip,tdm-data-pair: false + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f001c000 { + #sound-dai-cells = <0>; + compatible = "microchip,sam9x60-i2smcc"; + reg = <0xf001c000 0x100>; + interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(36))>, + <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(37))>; + dma-names = "tx", "rx"; + clocks = <&i2s_clk>, <&i2s_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml new file mode 100644 index 000000000000..9aa65c975c4e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-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. + +allOf: + - $ref: dai-common.yaml# + +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. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: value for DS line + enum: [0, 1] + - description: value for sampling edge + enum: [0, 1] + minItems: 1 + maxItems: 4 + uniqueItems: true + + microchip,startup-delay-us: + description: | + Specifies the delay in microseconds that needs to be applied after + enabling the PDMC microphones to avoid unwanted noise due to microphones + not being ready. + +required: + - compatible + - reg + - "#sound-dai-cells" + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - microchip,mic-pos + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/sound/microchip,pdmc.h> + + pdmc: sound@e1608000 { + compatible = "microchip,sama7g5-pdmc"; + reg = <0xe1608000 0x4000>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 68 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(37)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 68>, <&pmc PMC_TYPE_GCK 68>; + clock-names = "pclk", "gclk"; + microchip,mic-pos = <MCHP_PDMC_DS0 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS0 MCHP_PDMC_CLK_NEGATIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_NEGATIVE>; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml new file mode 100644 index 000000000000..2f43c684ab88 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Rx Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Receiver is a serial port + compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdifrx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdifrx: spdifrx@e1614000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdifrx"; + reg = <0xe1614000 0x4000>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(49)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 84>, <&pmc PMC_TYPE_GCK 84>; + clock-names = "pclk", "gclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml new file mode 100644 index 000000000000..4702c528700d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml @@ -0,0 +1,78 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller + +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. + +allOf: + - $ref: dai-common.yaml# + +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 + +unevaluatedProperties: 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/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt deleted file mode 100644 index 560762e0a168..000000000000 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ /dev/null @@ -1,146 +0,0 @@ -Mediatek AFE PCM controller for mt2701 - -Required properties: -- compatible: should be one of the followings. - - "mediatek,mt2701-audio" - - "mediatek,mt7622-audio" -- interrupts: should contain AFE and ASYS interrupts -- interrupt-names: should be "afe" and "asys" -- power-domains: should define the power domain -- clocks: Must contain an entry for each entry in clock-names - See ../clocks/clock-bindings.txt for details -- clock-names: should have these clock names: - "infra_sys_audio_clk", - "top_audio_mux1_sel", - "top_audio_mux2_sel", - "top_audio_a1sys_hp", - "top_audio_a2sys_hp", - "i2s0_src_sel", - "i2s1_src_sel", - "i2s2_src_sel", - "i2s3_src_sel", - "i2s0_src_div", - "i2s1_src_div", - "i2s2_src_div", - "i2s3_src_div", - "i2s0_mclk_en", - "i2s1_mclk_en", - "i2s2_mclk_en", - "i2s3_mclk_en", - "i2so0_hop_ck", - "i2so1_hop_ck", - "i2so2_hop_ck", - "i2so3_hop_ck", - "i2si0_hop_ck", - "i2si1_hop_ck", - "i2si2_hop_ck", - "i2si3_hop_ck", - "asrc0_out_ck", - "asrc1_out_ck", - "asrc2_out_ck", - "asrc3_out_ck", - "audio_afe_pd", - "audio_afe_conn_pd", - "audio_a1sys_pd", - "audio_a2sys_pd", - "audio_mrgif_pd"; -- assigned-clocks: list of input clocks and dividers for the audio system. - See ../clocks/clock-bindings.txt for details. -- assigned-clocks-parents: parent of input clocks of assigned clocks. -- assigned-clock-rates: list of clock frequencies of assigned clocks. - -Must be a subnode of MediaTek audsys device tree node. -See ../arm/mediatek/mediatek,audsys.txt for details about the parent node. - -Example: - - audsys: audio-subsystem@11220000 { - compatible = "mediatek,mt2701-audsys", "syscon"; - ... - - afe: audio-controller { - compatible = "mediatek,mt2701-audio"; - interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, - <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; - interrupt-names = "afe", "asys"; - power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; - - clocks = <&infracfg CLK_INFRA_AUDIO>, - <&topckgen CLK_TOP_AUD_MUX1_SEL>, - <&topckgen CLK_TOP_AUD_MUX2_SEL>, - <&topckgen CLK_TOP_AUD_48K_TIMING>, - <&topckgen CLK_TOP_AUD_44K_TIMING>, - <&topckgen CLK_TOP_AUD_K1_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K2_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K3_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K4_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K1_SRC_DIV>, - <&topckgen CLK_TOP_AUD_K2_SRC_DIV>, - <&topckgen CLK_TOP_AUD_K3_SRC_DIV>, - <&topckgen CLK_TOP_AUD_K4_SRC_DIV>, - <&topckgen CLK_TOP_AUD_I2S1_MCLK>, - <&topckgen CLK_TOP_AUD_I2S2_MCLK>, - <&topckgen CLK_TOP_AUD_I2S3_MCLK>, - <&topckgen CLK_TOP_AUD_I2S4_MCLK>, - <&audsys CLK_AUD_I2SO1>, - <&audsys CLK_AUD_I2SO2>, - <&audsys CLK_AUD_I2SO3>, - <&audsys CLK_AUD_I2SO4>, - <&audsys CLK_AUD_I2SIN1>, - <&audsys CLK_AUD_I2SIN2>, - <&audsys CLK_AUD_I2SIN3>, - <&audsys CLK_AUD_I2SIN4>, - <&audsys CLK_AUD_ASRCO1>, - <&audsys CLK_AUD_ASRCO2>, - <&audsys CLK_AUD_ASRCO3>, - <&audsys CLK_AUD_ASRCO4>, - <&audsys CLK_AUD_AFE>, - <&audsys CLK_AUD_AFE_CONN>, - <&audsys CLK_AUD_A1SYS>, - <&audsys CLK_AUD_A2SYS>, - <&audsys CLK_AUD_AFE_MRGIF>; - - clock-names = "infra_sys_audio_clk", - "top_audio_mux1_sel", - "top_audio_mux2_sel", - "top_audio_a1sys_hp", - "top_audio_a2sys_hp", - "i2s0_src_sel", - "i2s1_src_sel", - "i2s2_src_sel", - "i2s3_src_sel", - "i2s0_src_div", - "i2s1_src_div", - "i2s2_src_div", - "i2s3_src_div", - "i2s0_mclk_en", - "i2s1_mclk_en", - "i2s2_mclk_en", - "i2s3_mclk_en", - "i2so0_hop_ck", - "i2so1_hop_ck", - "i2so2_hop_ck", - "i2so3_hop_ck", - "i2si0_hop_ck", - "i2si1_hop_ck", - "i2si2_hop_ck", - "i2si3_hop_ck", - "asrc0_out_ck", - "asrc1_out_ck", - "asrc2_out_ck", - "asrc3_out_ck", - "audio_afe_pd", - "audio_afe_conn_pd", - "audio_a1sys_pd", - "audio_a2sys_pd", - "audio_mrgif_pd"; - - assigned-clocks = <&topckgen CLK_TOP_AUD_MUX1_SEL>, - <&topckgen CLK_TOP_AUD_MUX2_SEL>, - <&topckgen CLK_TOP_AUD_MUX1_DIV>, - <&topckgen CLK_TOP_AUD_MUX2_DIV>; - assigned-clock-parents = <&topckgen CLK_TOP_AUD1PLL_98M>, - <&topckgen CLK_TOP_AUD2PLL_90M>; - assigned-clock-rates = <0>, <0>, <49152000>, <45158400>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/mt6358.txt b/Documentation/devicetree/bindings/sound/mt6358.txt index 5465730013a1..fbe9e55c68f5 100644 --- a/Documentation/devicetree/bindings/sound/mt6358.txt +++ b/Documentation/devicetree/bindings/sound/mt6358.txt @@ -7,12 +7,20 @@ Must be a child node of PMIC wrapper. Required properties: -- compatible : "mediatek,mt6358-sound". +- compatible - "string" - One of: + "mediatek,mt6358-sound" + "mediatek,mt6366-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..23d411fc4200 --- /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 + +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-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 decaa013a07e..ecd46ed8eb98 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -1,13 +1,17 @@ -MT8183 with MT6358, TS3A227 and MAX98357 CODECS +MT8183 with MT6358, TS3A227, MAX98357, and RT1015 CODECS Required properties: -- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" +- 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: @@ -15,6 +19,7 @@ Example: 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/mt8186-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml new file mode 100644 index 000000000000..7fe85b08f9df --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml @@ -0,0 +1,175 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8186-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8186 + +maintainers: + - Jiaxin Yu <jiaxin.yu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8186-sound + + reg: + maxItems: 1 + + 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 + + clocks: + items: + - description: audio infra sys clock + - description: audio infra 26M clock + - description: audio top mux + - description: audio intbus mux + - description: mainpll 136.5M clock + - description: faud1 mux + - description: apll1 clock + - description: faud2 mux + - description: apll2 clock + - description: audio engen1 mux + - description: apll1_d8 22.5792M clock + - description: audio engen2 mux + - description: apll2_d8 24.576M clock + - description: i2s0 mclk mux + - description: i2s1 mclk mux + - description: i2s2 mclk mux + - description: i2s4 mclk mux + - description: tdm mclk mux + - description: i2s0_mck divider + - description: i2s1_mck divider + - description: i2s2_mck divider + - description: i2s4_mck divider + - description: tdm_mck divider + - description: audio hires mux + - description: 26M clock + + clock-names: + items: + - const: aud_infra_clk + - const: mtkaif_26m_clk + - const: top_mux_audio + - const: top_mux_audio_int + - const: top_mainpll_d2_d4 + - const: top_mux_aud_1 + - const: top_apll1_ck + - const: top_mux_aud_2 + - const: top_apll2_ck + - const: top_mux_aud_eng1 + - const: top_apll1_d8 + - const: top_mux_aud_eng2 + - const: top_apll2_d8 + - const: top_i2s0_m_sel + - const: top_i2s1_m_sel + - const: top_i2s2_m_sel + - const: top_i2s4_m_sel + - const: top_tdm_m_sel + - const: top_apll12_div0 + - const: top_apll12_div1 + - const: top_apll12_div2 + - const: top_apll12_div4 + - const: top_apll12_div_tdm + - const: top_mux_audio_h + - const: top_clk26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe: mt8186-afe-pcm@11210000 { + compatible = "mediatek,mt8186-sound"; + reg = <0x11210000 0x2000>; + interrupts = <GIC_SPI 169 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog 17>; //MT8186_TOPRGU_AUDIO_SW_RST + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + clocks = <&infracfg_ao 44>, //CLK_INFRA_AO_AUDIO + <&infracfg_ao 54>, //CLK_INFRA_AO_AUDIO_26M_BCLK + <&topckgen 15>, //CLK_TOP_AUDIO + <&topckgen 16>, //CLK_TOP_AUD_INTBUS + <&topckgen 70>, //CLK_TOP_MAINPLL_D2_D4 + <&topckgen 17>, //CLK_TOP_AUD_1 + <&apmixedsys 12>, //CLK_APMIXED_APLL1 + <&topckgen 18>, //CLK_TOP_AUD_2 + <&apmixedsys 13>, //CLK_APMIXED_APLL2 + <&topckgen 19>, //CLK_TOP_AUD_ENGEN1 + <&topckgen 101>, //CLK_TOP_APLL1_D8 + <&topckgen 20>, //CLK_TOP_AUD_ENGEN2 + <&topckgen 104>, //CLK_TOP_APLL2_D8 + <&topckgen 63>, //CLK_TOP_APLL_I2S0_MCK_SEL + <&topckgen 64>, //CLK_TOP_APLL_I2S1_MCK_SEL + <&topckgen 65>, //CLK_TOP_APLL_I2S2_MCK_SEL + <&topckgen 66>, //CLK_TOP_APLL_I2S4_MCK_SEL + <&topckgen 67>, //CLK_TOP_APLL_TDMOUT_MCK_SEL + <&topckgen 131>, //CLK_TOP_APLL12_CK_DIV0 + <&topckgen 132>, //CLK_TOP_APLL12_CK_DIV1 + <&topckgen 133>, //CLK_TOP_APLL12_CK_DIV2 + <&topckgen 134>, //CLK_TOP_APLL12_CK_DIV4 + <&topckgen 135>, //CLK_TOP_APLL12_CK_DIV_TDMOUT_M + <&topckgen 44>, //CLK_TOP_AUDIO_H + <&clk26m>; + clock-names = "aud_infra_clk", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_audio_int", + "top_mainpll_d2_d4", + "top_mux_aud_1", + "top_apll1_ck", + "top_mux_aud_2", + "top_apll2_ck", + "top_mux_aud_eng1", + "top_apll1_d8", + "top_mux_aud_eng2", + "top_apll2_d8", + "top_i2s0_m_sel", + "top_i2s1_m_sel", + "top_i2s2_m_sel", + "top_i2s4_m_sel", + "top_tdm_m_sel", + "top_apll12_div0", + "top_apll12_div1", + "top_apll12_div2", + "top_apll12_div4", + "top_apll12_div_tdm", + "top_mux_audio_h", + "top_clk26m_clk"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml new file mode 100644 index 000000000000..9853c11a1330 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.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/mt8186-mt6366-da7219-max98357.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek MT8186 with MT6366, DA7219 and MAX98357 ASoC sound card driver + +maintainers: + - Jiaxin Yu <jiaxin.yu@mediatek.com> + +description: + This binding describes the MT8186 sound card. + +properties: + compatible: + enum: + - mediatek,mt8186-mt6366-da7219-max98357-sound + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ASoC platform. + + headset-codec: + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + playback-codecs: + type: object + additionalProperties: false + properties: + sound-dai: + items: + - description: phandle of dp codec + - description: phandle of l channel speaker codec + - description: phandle of r channel speaker codec + minItems: 2 + required: + - sound-dai + + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ADSP platform. + + mediatek,dai-link: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - playback-codecs + +examples: + - | + + sound: mt8186-sound { + compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound"; + mediatek,platform = <&afe>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + + headset-codec { + sound-dai = <&da7219>; + }; + + playback-codecs { + sound-dai = <&anx_bridge_dp>, + <&max98357a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml new file mode 100644 index 000000000000..bdf7b0960533 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8186-mt6366-rt1019-rt5682s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek MT8186 with MT6366, RT1019 and RT5682S ASoC sound card driver + +maintainers: + - Jiaxin Yu <jiaxin.yu@mediatek.com> + +description: + This binding describes the MT8186 sound card. + +properties: + compatible: + enum: + - mediatek,mt8186-mt6366-rt1019-rt5682s-sound + - mediatek,mt8186-mt6366-rt5682s-max98360-sound + - mediatek,mt8186-mt6366-rt5650-sound + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ASoC platform. + + dmic-gpios: + maxItems: 1 + description: + dmic-gpios optional prop for switching between two DMICs. + Ex, the GPIO can control a MUX HW component to select + dmic clk and data form a Front or Rear dmic. + + headset-codec: + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + playback-codecs: + type: object + additionalProperties: false + properties: + sound-dai: + items: + - description: phandle of dp codec + - description: phandle of l channel speaker codec + - description: phandle of r channel speaker codec + minItems: 2 + required: + - sound-dai + + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ADSP platform. + + mediatek,dai-link: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - playback-codecs + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound: mt8186-sound { + compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound"; + mediatek,platform = <&afe>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on", + "aud_gpio_dmic_sec"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + pinctrl-2 = <&aud_gpio_dmic_sec>; + + dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>; + + headset-codec { + sound-dai = <&rt5682s>; + }; + + playback-codecs { + sound-dai = <&it6505dptx>, + <&rt1019p>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..064ef172bef4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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..7e50f5d65c8f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.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/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: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of HDMI codec. + + headset-codec: + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + speaker-codecs: + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 2 + items: + maxItems: 1 + required: + - sound-dai + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - speaker-codecs + +examples: + - | + + sound: mt8192-sound { + compatible = "mediatek,mt8192_mt6359_rt1015_rt5682"; + mediatek,platform = <&afe>; + mediatek,hdmi-codec = <&anx_bridge_dp>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + + headset-codec { + sound-dai = <&rt5682>; + }; + + speaker-codecs { + sound-dai = <&rt1015_l>, + <&rt1015_r>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml new file mode 100644 index 000000000000..5c8dba2b3a81 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml @@ -0,0 +1,200 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-in[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + "^mediatek,etdm-out[1-2]-cowork-source$": + $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 module. + 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..c1ddbf672ca3 --- /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/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt index cb8c07c81ce4..4f5dec5cb3c2 100644 --- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -6,9 +6,14 @@ Required properties: "marvell,kirkwood-audio" for Kirkwood platforms "marvell,dove-audio" for Dove platforms "marvell,armada370-audio" for Armada 370 platforms + "marvell,armada-380-audio" for Armada 38x platforms - reg: physical base address of the controller and length of memory mapped - region. + region (named "i2s_regs"). + With "marvell,armada-380-audio" two other regions are required: + first of those is dedicated for Audio PLL Configuration registers + (named "pll_regs") and the second one ("soc_ctrl") - for register + where one of exceptive I/O types (I2S or S/PDIF) is set. - interrupts: with "marvell,kirkwood-audio", the audio interrupt @@ -23,6 +28,13 @@ Required properties: "internal" for the internal clock "extclk" for the external clock +Optional properties: + +- spdif-mode: + Enable S/PDIF mode on Armada 38x SoC. Using this property + disables standard I2S I/O. Valid only with "marvell,armada-380-audio" + compatible string. + Example: i2s1: audio-controller@b4000 { 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/nau8540.txt b/Documentation/devicetree/bindings/sound/nau8540.txt deleted file mode 100644 index 307a76528320..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8540.txt +++ /dev/null @@ -1,16 +0,0 @@ -NAU85L40 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "nuvoton,nau8540" - - - reg : the I2C address of the device. - -Example: - -codec: nau8540@1c { - compatible = "nuvoton,nau8540"; - reg = <0x1c>; -}; diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt deleted file mode 100644 index 7deaa452b200..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8810.txt +++ /dev/null @@ -1,17 +0,0 @@ -NAU8810/NAU8812/NAU8814 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : One of "nuvoton,nau8810" or "nuvoton,nau8812" or - "nuvoton,nau8814" - - - reg : the I2C address of the device. - -Example: - -codec: nau8810@1a { - compatible = "nuvoton,nau8810"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt deleted file mode 100644 index a471d162d4e5..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8822.txt +++ /dev/null @@ -1,16 +0,0 @@ -NAU8822 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "nuvoton,nau8822" - - - reg : the I2C address of the device. - -Example: - -codec: nau8822@1a { - compatible = "nuvoton,nau8822"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/nau8824.txt b/Documentation/devicetree/bindings/sound/nau8824.txt deleted file mode 100644 index e0058b97e49a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8824.txt +++ /dev/null @@ -1,88 +0,0 @@ -Nuvoton NAU8824 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8824" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - 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,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 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,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - - - nuvoton,jack-eject-debounce: Jack ejection debounce time. - 0 - 0 ms - 1 - 1 ms - 2 - 10 ms - - -Example: - - headset: nau8824@1a { - compatible = "nuvoton,nau8824"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <0>; - nuvoton,sar-voltage = <6>; - nuvoton,sar-compare-time = <1>; - nuvoton,sar-sampling-time = <1>; - nuvoton,short-key-debounce = <0>; - nuvoton,jack-eject-debounce = <1>; - }; diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt deleted file mode 100644 index 388a7bc60b1f..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ /dev/null @@ -1,105 +0,0 @@ -Nuvoton NAU8825 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8825" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (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,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 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,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - 3 - 30 ms - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,crosstalk-enable: make crosstalk function enable if set. - - - clocks: list of phandle and clock specifier pairs according to common clock bindings for the - clocks described in clock-names - - clock-names: should include "mclk" for the MCLK master clock - -Example: - - headset: nau8825@1a { - compatible = "nuvoton,nau8825"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) 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>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <1>; - nuvoton,sar-voltage = <0>; - nuvoton,sar-compare-time = <0>; - nuvoton,sar-sampling-time = <0>; - nuvoton,short-key-debounce = <2>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-enable; - - clock-names = "mclk"; - clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml new file mode 100644 index 000000000000..24006e9dc501 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8315.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8315/NAU8318 Mono Class-D Amplifier + +maintainers: + - David Lin <CTLIN0@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8315 + - nuvoton,nau8318 + + '#sound-dai-cells': + const: 0 + + enable-gpios: + maxItems: 1 + description: + GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + codec { + compatible = "nuvoton,nau8315"; + #sound-dai-cells = <0>; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8540.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8540.yaml new file mode 100644 index 000000000000..7ccfbb8d8b04 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8540.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8540.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Nuvoton Technology Corporation NAU85L40 Audio CODEC + +maintainers: + - John Hsu <KCHSU0@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8540 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1c { + compatible = "nuvoton,nau8540"; + reg = <0x1c>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml new file mode 100644 index 000000000000..d9696f6c75ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8810.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8810/NAU8812/NAU8814 audio CODEC + +maintainers: + - David Lin <CTLIN0@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8810 + - nuvoton,nau8812 + - nuvoton,nau8814 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8810"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..054b53954ac3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,141 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,key-enable: + description: handles key press detection. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + nuvoton,dmic-slew-rate: + description: The range 0 to 7 represents the speed of DMIC slew rate. + The lowest value 0 means the slowest rate and the highest value + 7 means the fastest rate. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,key-enable; + nuvoton,left-input-single-end; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + nuvoton,dmic-slew-rate = <0>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml new file mode 100644 index 000000000000..cb8182bbc491 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8822.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8822 audio CODEC + +description: | + 24 bit stereo audio codec with speaker driver. + This device supports I2C/SPI. + +maintainers: + - David Lin <CTLIN0@nuvoton.com> + +properties: + compatible: + enum: + - nuvoton,nau8822 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + nuvoton,spk-btl: + description: + If set, configure the two loudspeaker outputs as a Bridge Tied Load output + to drive a high power external loudspeaker. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml new file mode 100644 index 000000000000..3dbf438c3841 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8824.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8824 audio CODEC + +maintainers: + - John Hsu <KCHSU0@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8824 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 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 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 8 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 8 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 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 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500ns + - 1 # 1us + - 2 # 2us + - 3 # 4us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2us + - 1 # 4us + - 2 # 8us + - 3 # 16us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + default: 0 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Jack ejection debounce time. + enum: + - 0 # 0 ms + - 1 # 1 ms + - 2 # 10 ms + default: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,sar-threshold-num = <4>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml new file mode 100644 index 000000000000..a54f194a0b49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml @@ -0,0 +1,239 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8825 audio CODEC + +maintainers: + - John Hsu <KCHSU0@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8825 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: + Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: + Enable JKDET pin pull. + If set - pin pull enabled, otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: + Pull-up JKDET pin. + If set then JKDET pin is pull up, otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 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 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 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 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500 ns + - 1 # 1 us + - 2 # 2 us + - 3 # 4 us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2 us + - 1 # 4 us + - 2 # 8 us + - 3 # 16 us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + - 3 # 30 ms + default: 3 + + nuvoton,jack-insert-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms. + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms + maximum: 7 + default: 0 + + nuvoton,crosstalk-enable: + description: + make crosstalk function enable if set. + type: boolean + + nuvoton,adcout-drive-strong: + description: + make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + type: boolean + + nuvoton,adc-delay-ms: + description: + Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 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>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_pmc 1>; + }; + }; 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..96f2f927a6f5 --- /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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headset Stereophone + - Int Spk + - Headset Mic + - Digital Mic + + # CODEC Pins + - SPKOUT + - SPKOUTN + - MICBIAS1 + - MIC1 + - HPR + - HPL + - DMICDAT + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "Headset Mic", "MICBIAS1", + "MIC1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL", + "DMICDAT", "Digital Mic"; + + nvidia,i2s-controller = <&i2s>; + nvidia,audio-codec = <&codec>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml new file mode 100644 index 000000000000..2588589ad62d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,ear-sel-gpios: + maxItems: 1 + description: The GPIO that switch between the microphones + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + + nvidia,coupled-mic-hp-det: + type: boolean + description: The Mic detect GPIO is viable only if HP detect GPIO is active + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml new file mode 100644 index 000000000000..b4bee466d67a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -0,0 +1,199 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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-max9808x.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml new file mode 100644 index 000000000000..241d20f3aad0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-max9808x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with MAX9808x CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98088(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-max98088 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98089(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-max98089 + + 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 + - Int Spk + - Headphone Jack + - Earpiece + - Headset Mic + - Internal Mic 1 + - Internal Mic 2 + + # CODEC Pins + - HPL + - HPR + - SPKL + - SPKR + - RECL + - RECR + - INA1 + - INA2 + - INB1 + - INB2 + - MIC1 + - MIC2 + - MICBIAS + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + #include <dt-bindings/soc/tegra-pmc.h> + sound { + compatible = "lg,tegra-audio-max98089-p895", + "nvidia,tegra-audio-max98089"; + nvidia,model = "LG Optimus Vu MAX98089"; + + nvidia,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Int Spk", "SPKL", + "Int Spk", "SPKR", + "Earpiece", "RECL", + "Earpiece", "RECR", + "INA1", "Headset Mic", + "MIC1", "MICBIAS", + "MICBIAS", "Internal Mic 1", + "MIC2", "Internal Mic 2"; + + nvidia,i2s-controller = <&tegra_i2s0>; + nvidia,audio-codec = <&codec>; + + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_pmc TEGRA_PMC_CLK_OUT_1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; 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..4d912458b18b --- /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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphones + - Speakers + - Mic Jack + - Int Mic + + # CODEC Pins + - MIC1 + - MIC2 + - DMICL + - DMICR + - IN1 + - IN2 + - IN3 + - IN4 + - IN5 + - IN6 + - IN12 + - IN34 + - IN56 + - HPL + - HPR + - SPKL + - SPKR + - RCVL + - RCVR + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra124-car.h> + + sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml new file mode 100644 index 000000000000..0c8067c3b056 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5631.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5631 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5631(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5631 + + 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 + - Int Spk + - Headphone Jack + - Mic Jack + - Int Mic + + # CODEC Pins + - MIC1 + - MIC2 + - AXIL + - AXIR + - MONOIN_RXN + - MONOIN_RXP + - DMIC + - MIC Bias1 + - MIC Bias2 + - MONO_IN + - AUXO1 + - AUXO2 + - SPOL + - SPOR + - HPOL + - HPOR + - MONO + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + #include <dt-bindings/soc/tegra-pmc.h> + sound { + compatible = "asus,tegra-audio-rt5631-tf700t", + "nvidia,tegra-audio-rt5631"; + nvidia,model = "Asus Transformer Infinity TF700T RT5631"; + + nvidia,audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR", + "Int Spk", "SPOL", + "Int Spk", "SPOR", + "MIC1", "MIC Bias1", + "MIC Bias1", "Mic Jack", + "DMIC", "Int Mic"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5631>; + + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_pmc TEGRA_PMC_CLK_OUT_1>; + 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..2638592435b2 --- /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..09e1d0b18d27 --- /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..e5bc6a6ade24 --- /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..3323d6a438f5 --- /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 bbd581a8c5bc..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ /dev/null @@ -1,62 +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 - * Int Mic - -- 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 -- nvidia,headset : The Mic Jack represents state of the headset microphone pin - -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..1be25ce4514b --- /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..397306b8800d --- /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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphone + - LineIn + - Mic + + # CODEC Pins + - MONOOUT + - HPOUTL + - HPOUTR + - LOUT2 + - ROUT2 + - OUT3 + - LINEINL + - LINEINR + - PHONE + - PCBEEP + - MIC1 + - MIC2 + - Mic Bias + +required: + - nvidia,ac97-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm9712-colibri_t20", + "nvidia,tegra-audio-wm9712"; + nvidia,model = "Toradex Colibri T20"; + + nvidia,audio-routing = + "Headphone", "HPOUTL", + "Headphone", "HPOUTR", + "LineIn", "LINEINL", + "LineIn", "LINEINR", + "Mic", "MIC1"; + + nvidia,ac97-controller = <&ac97>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml new file mode 100644 index 000000000000..e15f387c4c29 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra186-asrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra186 ASRC + +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: dai-common.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]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + asrc@2910000 { + compatible = "nvidia,tegra186-asrc"; + reg = <0x2910000 0x2000>; + sound-name-prefix = "ASRC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml new file mode 100644 index 000000000000..e1362c77472b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra186-dspk.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra186 DSPK Controller + +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: dai-common.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# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 I2S Controller + +description: | + The I2S Controller streams synchronous serial audio data between system + memory and an external audio device. The controller supports the I2S Left + Justified Mode, Right Justified Mode, and DSP mode formats. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-i2s + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: i2s + + interrupts: + maxItems: 1 + + clocks: + minItems: 1 + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = <45>; + clocks = <&tegra_car 11>; + resets = <&tegra_car 11>; + reset-names = "i2s"; + dmas = <&apbdma 21>, <&apbdma 21>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml new file mode 100644 index 000000000000..dc76a4dc0ed2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nvidia,tegra20-spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 2 + + clock-names: + items: + - const: out + - const: in + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + spdif@70002400 { + compatible = "nvidia,tegra20-spdif"; + reg = <0x70002400 0x200>; + interrupts = <77>; + clocks = <&clk 99>, <&clk 98>; + clock-names = "out", "in"; + resets = <&rst 10>; + dmas = <&apbdma 3>, <&apbdma 3>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml new file mode 100644 index 000000000000..15ab40aeab1e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml @@ -0,0 +1,129 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-admaif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 ADMAIF + +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)$" + description: + Should be "rx1", "rx2" ... "rx20" for DMA Rx channel + Should be "tx1", "tx2" ... "tx20" for DMA Tx channel + minItems: 1 + maxItems: 40 + +required: + - compatible + - reg + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml new file mode 100644 index 000000000000..e4c871797fa6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-adx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 ADX + +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: dai-common.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]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + adx@702d3800 { + compatible = "nvidia,tegra210-adx"; + reg = <0x702d3800 0x100>; + sound-name-prefix = "ADX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml new file mode 100644 index 000000000000..c4abac81f207 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml @@ -0,0 +1,196 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ahub.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 AHUB + +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": + enum: [ 1, 2 ] + + "#size-cells": + enum: [ 1, 2 ] + + 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 + $ref: nvidia,tegra186-asrc.yaml# + + '^processing-engine@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-ope.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..021b72546ba4 --- /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 + +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: dai-common.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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF output port + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amx@702d3000 { + compatible = "nvidia,tegra210-amx"; + reg = <0x702d3000 0x100>; + sound-name-prefix = "AMX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml new file mode 100644 index 000000000000..bff551c35da7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 DMIC Controller + +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: dai-common.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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC DAP (Digital Audio Port) interface which can be connected + to external audio codec for capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + dmic@702d4000 { + compatible = "nvidia,tegra210-dmic"; + reg = <0x702d4000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + clock-names = "dmic"; + assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <3072000>; + sound-name-prefix = "DMIC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml new file mode 100644 index 000000000000..a82f11fb6c9a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 I2S Controller + +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: dai-common.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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback or capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + reg = <0x702d1000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + sound-name-prefix = "I2S1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml new file mode 100644 index 000000000000..5b9198602fc6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mbdrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 MBDRC + +description: + The Multi Band Dynamic Range Compressor (MBDRC) is part of Output + Processing Engine (OPE) which interfaces with Audio Hub (AHUB) via + Audio Client Interface (ACIF). MBDRC can be used as a traditional + single full band or a dual band or a multi band dynamic processor. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-mbdrc + - items: + - enum: + - nvidia,tegra234-mbdrc + - nvidia,tegra194-mbdrc + - nvidia,tegra186-mbdrc + - const: nvidia,tegra210-mbdrc + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + dynamic-range-compressor@702d8200 { + compatible = "nvidia,tegra210-mbdrc"; + reg = <0x702d8200 0x200>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml new file mode 100644 index 000000000000..049898f02e85 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mixer.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 Mixer + +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: dai-common.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]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amixer@702dbb00 { + compatible = "nvidia,tegra210-amixer"; + reg = <0x702dbb00 0x800>; + sound-name-prefix = "MIXER1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml new file mode 100644 index 000000000000..d0280d8aa3af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mvc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 MVC + +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: dai-common.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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + mvc@702da000 { + compatible = "nvidia,tegra210-mvc"; + reg = <0x702da000 0x200>; + sound-name-prefix = "MVC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml new file mode 100644 index 000000000000..9017fb6d575d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ope.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 OPE + +description: + The Output Processing Engine (OPE) is one of the AHUB client. It has + PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) + sub blocks for data processing. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-ope + - items: + - enum: + - nvidia,tegra234-ope + - nvidia,tegra194-ope + - nvidia,tegra186-ope + - const: nvidia,tegra210-ope + + reg: + maxItems: 1 + + "#address-cells": + enum: [ 1, 2 ] + + "#size-cells": + enum: [ 1, 2 ] + + ranges: true + + sound-name-prefix: + pattern: "^OPE[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: + OPE ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: + OPE ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +patternProperties: + '^equalizer@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-peq.yaml# + + '^dynamic-range-compressor@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mbdrc.yaml# + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + processing-engine@702d8000 { + compatible = "nvidia,tegra210-ope"; + reg = <0x702d8000 0x100>; + sound-name-prefix = "OPE1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml new file mode 100644 index 000000000000..1e373c49d639 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-peq.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 PEQ + +description: + The Parametric Equalizer (PEQ) is a cascade of biquad filters with + each filter tuned based on certain parameters. It can be used to + equalize the irregularities in the speaker frequency response. + PEQ sits inside Output Processing Engine (OPE) which interfaces + with Audio Hub (AHUB) via Audio Client Interface (ACIF). + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-peq + - items: + - enum: + - nvidia,tegra234-peq + - nvidia,tegra194-peq + - nvidia,tegra186-peq + - const: nvidia,tegra210-peq + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + equalizer@702d8100 { + compatible = "nvidia,tegra210-peq"; + reg = <0x702d8100 0x100>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml new file mode 100644 index 000000000000..185ca0be4f02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-sfc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 SFC + +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: dai-common.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: + $ref: /schemas/types.yaml#/definitions/string + description: | + The user-visible name of this sound complex. If this property is + not specified then boards can use default name provided in hda driver. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra124-car-common.h> + #include<dt-bindings/interrupt-controller/arm-gic.h> + + hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hda2codec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>, /* hda2hdmi */ + <&tegra_car 111>; /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hda2codec_2x"; + nvidia,model = "jetson-tk1-hda"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml new file mode 100644 index 000000000000..df26248573ad --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,tfa9879.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP TFA9879 class-D audio amplifier + +maintainers: + - Peter Rosin <peda@axentia.se> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nxp,tfa9879 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + i2c1 { + #address-cells = <1>; + #size-cells = <0>; + amplifier@6c { + compatible = "nxp,tfa9879"; + reg = <0x6c>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c1>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml new file mode 100644 index 000000000000..fd2415e231eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nxp,tfa9890 + - nxp,tfa9895 + - nxp,tfa9897 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + rcv-gpios: + description: optional GPIO to be asserted when receiver mode is enabled. + + sound-name-prefix: true + + vddd-supply: + description: regulator phandle for the VDDD power supply. + +if: + not: + properties: + compatible: + const: nxp,tfa9897 +then: + properties: + rcv-gpios: false + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@34 { + compatible = "nxp,tfa9895"; + reg = <0x34>; + sound-name-prefix = "Speaker Left"; + #sound-dai-cells = <0>; + }; + audio-codec@36 { + compatible = "nxp,tfa9895"; + reg = <0x36>; + sound-name-prefix = "Speaker Right"; + #sound-dai-cells = <0>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + speaker_codec_top: audio-codec@34 { + compatible = "nxp,tfa9897"; + reg = <0x34>; + vddd-supply = <&pm8916_l6>; + rcv-gpios = <&msmgpio 50 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&speaker_top_default>; + sound-name-prefix = "Speaker Top"; + #sound-dai-cells = <0>; + }; + + speaker_codec_bottom: audio-codec@36 { + compatible = "nxp,tfa9897"; + reg = <0x36>; + vddd-supply = <&pm8916_l6>; + rcv-gpios = <&msmgpio 111 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&speaker_bottom_default>; + sound-name-prefix = "Speaker Bottom"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/option,gtm601.yaml b/Documentation/devicetree/bindings/sound/option,gtm601.yaml new file mode 100644 index 000000000000..ff813d97fc59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/option,gtm601.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/option,gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel@puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - description: Broadmobi BM818 (48Khz stereo) + items: + - const: broadmobi,bm818 + - const: option,gtm601 + - description: GTM601 (8kHz mono) + const: option,gtm601 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + +allOf: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + codec { + compatible = "option,gtm601"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index 3aae3b41bd8e..77006a4aec4a 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -1,12 +1,12 @@ -PCM512x audio CODECs +PCM512x and TAS575x audio CODECs/amplifiers These devices support both I2C and SPI (configured with pin strapping -on the board). +on the board). The TAS575x devices only support I2C. Required properties: - - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or - "ti,pcm5142" + - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141", + "ti,pcm5142", "ti,tas5754" or "ti,tas5756" - reg : the I2C address of the device for I2C, the chip select number for SPI. @@ -25,6 +25,7 @@ Optional properties: through <6>. The device will be configured for clock input on the given pll-in pin and PLL output on the given pll-out pin. An external connection from the pll-out pin to the SCLK pin is assumed. + Caution: the TAS-desvices only support gpios 1,2 and 3 Examples: 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 32c2cdb3d32f..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ /dev/null @@ -1,79 +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" -- #address-cells : Must be 1 -- #size-cells : Must be 0 - - - -Optional properties: - -- qcom,adsp : Phandle for the audio DSP node - -By default, the driver uses up to 4 MI2S SD lines, for a total of 8 channels. -The SD lines to use can be configured by adding subnodes for each of the DAIs. - -Required properties for each DAI (represented by a subnode): -- reg : Must be one of the DAI IDs - (usually part of dt-bindings header) -- qcom,playback-sd-lines: List of serial data lines to use for playback - Each SD line should be represented by a number from 0-3. -- qcom,capture-sd-lines : List of serial data lines to use for capture - Each SD line should be represented by a number from 0-3. - -Note that adding a subnode changes the default to "no lines configured", -so both playback and capture lines should be configured when a subnode is added. - -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>; - - #address-cells = <1>; - #size-cells = <0>; - - /* Optional to set different MI2S SD lines */ - dai@3 { - reg = <MI2S_QUATERNARY>; - qcom,playback-sd-lines = <0 1>; - }; -}; 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..3a559bd07a79 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -0,0 +1,290 @@ +# 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 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + - Rohit kumar <quic_rohkumar@quicinc.com> + +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: 1 + maxItems: 6 + description: LPAIF core registers + + reg-names: + minItems: 1 + maxItems: 6 + + clocks: + minItems: 3 + maxItems: 10 + + clock-names: + minItems: 1 + maxItems: 10 + + interrupts: + minItems: 1 + maxItems: 4 + description: LPAIF DMA buffer interrupt + + interrupt-names: + minItems: 1 + maxItems: 4 + + qcom,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle for the audio DSP node + + iommus: + minItems: 2 + maxItems: 3 + description: Phandle to apps_smmu node with sid mask + + power-domains: + maxItems: 1 + + power-domain-names: + maxItems: 1 + + required-opps: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]+$": + 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' + +unevaluatedProperties: false + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: qcom,lpass-cpu + + then: + properties: + clocks: + maxItems: 3 + 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: + clocks: + minItems: 7 + maxItems: 7 + 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: + clocks: + minItems: 6 + maxItems: 6 + clock-names: + items: + - const: pcnoc-sway-clk + - const: audio-core + - const: mclk0 + - const: pcnoc-mport-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + reg: + minItems: 2 + maxItems: 2 + reg-names: + items: + - const: lpass-hdmiif + - const: lpass-lpaif + interrupts: + minItems: 2 + maxItems: 2 + interrupt-names: + items: + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + required: + - iommus + - power-domains + + - if: + properties: + compatible: + contains: + const: qcom,sc7280-lpass-cpu + + then: + properties: + clocks: + minItems: 10 + maxItems: 10 + clock-names: + items: + - 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 + - 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 + reg: + minItems: 6 + maxItems: 6 + reg-names: + items: + - 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 + interrupts: + minItems: 4 + maxItems: 4 + interrupt-names: + items: + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - const: lpass-irq-vaif + - const: lpass-irq-rxtxif + power-domain-names: + 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..b8540b30741e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.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/qcom,lpass-rx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) RX Macro audio codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + - qcom,sm8550-lpass-rx-macro + - qcom,sc8280xp-lpass-rx-macro + - items: + - enum: + - qcom,sm8650-lpass-rx-macro + - qcom,x1e80100-lpass-rx-macro + - const: qcom,sm8550-lpass-rx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 3 + maxItems: 5 + + clock-names: + minItems: 3 + maxItems: 5 + + clock-output-names: + maxItems: 1 + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-rx-macro + then: + properties: + clock-names: + oneOf: + - items: # for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: # for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sc8280xp-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-rx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: fsgen + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3200000 { + compatible = "qcom,sm8250-lpass-rx-macro"; + reg = <0x3200000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 0>, + <&audiocc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml new file mode 100644 index 000000000000..3e2ae16c6aba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -0,0 +1,160 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-tx-macro + - qcom,sm6115-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + - qcom,sm8450-lpass-tx-macro + - qcom,sm8550-lpass-tx-macro + - qcom,sc8280xp-lpass-tx-macro + - items: + - enum: + - qcom,sm8650-lpass-tx-macro + - qcom,x1e80100-lpass-tx-macro + - const: qcom,sm8550-lpass-tx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 3 + maxItems: 5 + + clock-names: + minItems: 3 + maxItems: 5 + + clock-output-names: + maxItems: 1 + + 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" + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-tx-macro + then: + properties: + 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 + + - if: + properties: + compatible: + enum: + - qcom,sc8280xp-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + - qcom,sm8450-lpass-tx-macro + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sm6115-lpass-tx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: npl + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-tx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: fsgen + +unevaluatedProperties: 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..6b483fa3c428 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -0,0 +1,154 @@ +# 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 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-va-macro + - qcom,sm8250-lpass-va-macro + - qcom,sm8450-lpass-va-macro + - qcom,sm8550-lpass-va-macro + - qcom,sc8280xp-lpass-va-macro + - items: + - enum: + - qcom,sm8650-lpass-va-macro + - qcom,x1e80100-lpass-va-macro + - const: qcom,sm8550-lpass-va-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 1 + maxItems: 4 + + clock-names: + minItems: 1 + maxItems: 4 + + clock-output-names: + maxItems: 1 + + 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" + - clock-names + - clocks + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + const: qcom,sc7280-lpass-va-macro + then: + properties: + clocks: + maxItems: 1 + clock-names: + items: + - const: mclk + + - if: + properties: + compatible: + contains: + const: qcom,sm8250-lpass-va-macro + then: + properties: + clocks: + minItems: 3 + maxItems: 3 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + + - if: + properties: + compatible: + contains: + enum: + - qcom,sc8280xp-lpass-va-macro + - qcom,sm8450-lpass-va-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: npl + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-va-macro + then: + properties: + clocks: + minItems: 3 + maxItems: 3 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3370000 { + compatible = "qcom,sm8250-lpass-va-macro"; + reg = <0x3370000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; + clock-names = "mclk", "macro", "dcodec"; + clock-output-names = "fsgen"; + qcom,dmic-sample-rate = <600000>; + vdd-micb-supply = <&vreg_s4a_1p8>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml new file mode 100644 index 000000000000..06b5f7be3608 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-wsa-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sm8550-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + - items: + - enum: + - qcom,sm8650-lpass-wsa-macro + - qcom,x1e80100-lpass-wsa-macro + - const: qcom,sm8550-lpass-wsa-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 4 + maxItems: 6 + + clock-names: + minItems: 4 + maxItems: 6 + + clock-output-names: + maxItems: 1 + + 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" + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sm8250-lpass-wsa-macro + then: + properties: + clocks: + minItems: 6 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: va + - const: fsgen + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-wsa-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: fsgen + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/qcom,sm8250-lpass-aoncc.h> + #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>, + <&aoncc LPASS_CDC_VA_MCLK>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt deleted file mode 100644 index e7d17dda55db..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt +++ /dev/null @@ -1,101 +0,0 @@ -msm8916 analog audio CODEC - -Bindings for codec Analog IP which is integrated in pmic pm8916, - -## Bindings for codec core on pmic: - -Required properties - - compatible = "qcom,pm8916-wcd-analog-codec"; - - reg: represents the slave base address provided to the peripheral. - - interrupts: List of interrupts in given SPMI peripheral. - - interrupt-names: Names specified to above list of interrupts in same - order. List of supported interrupt names are: - "cdc_spk_cnp_int" - Speaker click and pop interrupt. - "cdc_spk_clip_int" - Speaker clip interrupt. - "cdc_spk_ocp_int" - Speaker over current protect interrupt. - "mbhc_ins_rem_det1" - jack insert removal detect interrupt 1. - "mbhc_but_rel_det" - button release interrupt. - "mbhc_but_press_det" - button press event - "mbhc_ins_rem_det" - jack insert removal detect interrupt. - "mbhc_switch_int" - multi button headset interrupt. - "cdc_ear_ocp_int" - Earphone over current protect interrupt. - "cdc_hphr_ocp_int" - Headphone R over current protect interrupt. - "cdc_hphl_ocp_det" - Headphone L over current protect interrupt. - "cdc_ear_cnp_int" - earphone cnp interrupt. - "cdc_hphr_cnp_int" - hphr click and pop interrupt. - "cdc_hphl_cnp_int" - hphl click and pop interrupt. - - - clocks: Handle to mclk. - - clock-names: should be "mclk" - - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node. - - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node. - - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node. - -Optional Properties: - - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons - detection on headset when the mbhc is powered up - by internal current source, this is a low power. - - qcom,mbhc-vthreshold-high: Array of 5 thresold voltages in mV for 5 buttons - detection on headset when mbhc is powered up - from micbias. -- qcom,micbias-lvl: Voltage (mV) for Mic Bias -- qcom,hphl-jack-type-normally-open: boolean, present if hphl pin on jack is a - NO (Normally Open). If not specified, then - its assumed that hphl pin on jack is NC - (Normally Closed). -- qcom,gnd-jack-type-normally-open: boolean, present if gnd pin on jack is - NO (Normally Open). If not specified, then - its assumed that gnd pin on jack is NC - (Normally Closed). -- qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor - connected. -- qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor - connected. - -Example: - -spmi_bus { - ... - audio-codec@f000{ - compatible = "qcom,pm8916-wcd-analog-codec"; - reg = <0xf000 0x200>; - reg-names = "pmic-codec-core"; - clocks = <&gcc GCC_CODEC_DIGCODEC_CLK>; - clock-names = "mclk"; - qcom,mbhc-vthreshold-low = <75 150 237 450 500>; - qcom,mbhc-vthreshold-high = <75 150 237 450 500>; - interrupt-parent = <&spmi_bus>; - interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, - <0x1 0xf0 0x1 IRQ_TYPE_NONE>, - <0x1 0xf0 0x2 IRQ_TYPE_NONE>, - <0x1 0xf0 0x3 IRQ_TYPE_NONE>, - <0x1 0xf0 0x4 IRQ_TYPE_NONE>, - <0x1 0xf0 0x5 IRQ_TYPE_NONE>, - <0x1 0xf0 0x6 IRQ_TYPE_NONE>, - <0x1 0xf0 0x7 IRQ_TYPE_NONE>, - <0x1 0xf1 0x0 IRQ_TYPE_NONE>, - <0x1 0xf1 0x1 IRQ_TYPE_NONE>, - <0x1 0xf1 0x2 IRQ_TYPE_NONE>, - <0x1 0xf1 0x3 IRQ_TYPE_NONE>, - <0x1 0xf1 0x4 IRQ_TYPE_NONE>, - <0x1 0xf1 0x5 IRQ_TYPE_NONE>; - interrupt-names = "cdc_spk_cnp_int", - "cdc_spk_clip_int", - "cdc_spk_ocp_int", - "mbhc_ins_rem_det1", - "mbhc_but_rel_det", - "mbhc_but_press_det", - "mbhc_ins_rem_det", - "mbhc_switch_int", - "cdc_ear_ocp_int", - "cdc_hphr_ocp_int", - "cdc_hphl_ocp_det", - "cdc_ear_cnp_int", - "cdc_hphr_cnp_int", - "cdc_hphl_cnp_int"; - vdd-cdc-io-supply = <&pm8916_l5>; - vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>; - vdd-micbias-supply = <&pm8916_l13>; - #sound-dai-cells = <1>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml b/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml new file mode 100644 index 000000000000..94e7a1860977 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.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,pm8916-wcd-analog-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm PM8916 WCD Analog Audio Codec + +maintainers: + - Konrad Dybcio <konradybcio@kernel.org> + +description: + The analog WCD audio codec found on Qualcomm PM8916 PMIC. + +properties: + compatible: + const: qcom,pm8916-wcd-analog-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 14 + + interrupt-names: + items: + - const: cdc_spk_cnp_int + - const: cdc_spk_clip_int + - const: cdc_spk_ocp_int + - const: mbhc_ins_rem_det1 + - const: mbhc_but_rel_det + - const: mbhc_but_press_det + - const: mbhc_ins_rem_det + - const: mbhc_switch_int + - const: cdc_ear_ocp_int + - const: cdc_hphr_ocp_int + - const: cdc_hphl_ocp_det + - const: cdc_ear_cnp_int + - const: cdc_hphr_cnp_int + - const: cdc_hphl_cnp_int + + vdd-cdc-io-supply: + description: 1.8V buck supply + + vdd-cdc-tx-rx-cx-supply: + description: 1.8V SIDO buck supply + + vdd-micbias-supply: + description: micbias supply + + qcom,mbhc-vthreshold-low: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Array of 5 threshold voltages in mV for 5-button detection on + headset when MBHC is powered by an internal current source. + minItems: 5 + maxItems: 5 + + qcom,mbhc-vthreshold-high: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Array of 5 threshold voltages in mV for 5-button detection on + headset when MBHC is powered from micbias. + minItems: 5 + maxItems: 5 + + qcom,micbias-lvl: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Voltage (mV) for Mic Bias + + qcom,hphl-jack-type-normally-open: + type: boolean + description: + True if the HPHL pin on the jack is NO (Normally Open), false if it's + NC (Normally Closed). + + qcom,gnd-jack-type-normally-open: + type: boolean + description: + True if the GND pin on the jack is NO (Normally Open), false if it's + NC (Normally Closed). + + qcom,micbias1-ext-cap: + type: boolean + description: + True if micbias1 has an external capacitor. + + qcom,micbias2-ext-cap: + type: boolean + description: + True if micbias2 has an external capacitor. + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/spmi/spmi.h> + + pmic@1 { + compatible = "qcom,pm8916", "qcom,spmi-pmic"; + reg = <0x1 SPMI_USID>; + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@f000 { + compatible = "qcom,pm8916-wcd-analog-codec"; + reg = <0xf000>; + qcom,mbhc-vthreshold-low = <75 150 237 450 500>; + qcom,mbhc-vthreshold-high = <75 150 237 450 500>; + interrupt-parent = <&spmi_bus>; + interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, + <0x1 0xf0 0x1 IRQ_TYPE_NONE>, + <0x1 0xf0 0x2 IRQ_TYPE_NONE>, + <0x1 0xf0 0x3 IRQ_TYPE_NONE>, + <0x1 0xf0 0x4 IRQ_TYPE_NONE>, + <0x1 0xf0 0x5 IRQ_TYPE_NONE>, + <0x1 0xf0 0x6 IRQ_TYPE_NONE>, + <0x1 0xf0 0x7 IRQ_TYPE_NONE>, + <0x1 0xf1 0x0 IRQ_TYPE_NONE>, + <0x1 0xf1 0x1 IRQ_TYPE_NONE>, + <0x1 0xf1 0x2 IRQ_TYPE_NONE>, + <0x1 0xf1 0x3 IRQ_TYPE_NONE>, + <0x1 0xf1 0x4 IRQ_TYPE_NONE>, + <0x1 0xf1 0x5 IRQ_TYPE_NONE>; + interrupt-names = "cdc_spk_cnp_int", + "cdc_spk_clip_int", + "cdc_spk_ocp_int", + "mbhc_ins_rem_det1", + "mbhc_but_rel_det", + "mbhc_but_press_det", + "mbhc_ins_rem_det", + "mbhc_switch_int", + "cdc_ear_ocp_int", + "cdc_hphr_ocp_int", + "cdc_hphl_ocp_det", + "cdc_ear_cnp_int", + "cdc_hphr_cnp_int", + "cdc_hphl_cnp_int"; + vdd-cdc-io-supply = <&pm8916_l5>; + vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>; + vdd-micbias-supply = <&pm8916_l13>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml b/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml new file mode 100644 index 000000000000..3f11d2e183e1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml @@ -0,0 +1,39 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6adm-routing.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Device Manager (Q6ADM) routing + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Qualcomm Audio Device Manager (Q6ADM) routing node represents routing + specific configuration. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - qcom,q6adm-routing + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + routing { + compatible = "qcom,q6adm-routing"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt deleted file mode 100644 index 15c353a20de8..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt +++ /dev/null @@ -1,39 +0,0 @@ -Qualcomm Audio Device Manager (Q6ADM) binding - -Q6ADM is one of the APR audio service on Q6DSP. -Please refer to qcom,apr.txt for details of the coommon apr service bindings -used by the apr service device. - -- but must contain the following property: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6adm-v<MAJOR-NUMBER>.<MINOR-NUMBER>". - Or "qcom,q6adm" where the version number can be queried - from DSP. - example "qcom,q6adm-v2.0" - - -= ADM routing -"routing" subnode of the ADM node represents adm routing specific configuration - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6adm-routing". - -- #sound-dai-cells - Usage: required - Value type: <u32> - Definition: Must be 0 - -= EXAMPLE -apr-service@8 { - compatible = "qcom,q6adm"; - reg = <APR_SVC_ADM>; - q6routing: routing { - compatible = "qcom,q6adm-routing"; - #sound-dai-cells = <0>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6adm.yaml new file mode 100644 index 000000000000..fe14a97ea616 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6adm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Device Manager (Q6ADM) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6adm + + routing: + type: object + $ref: /schemas/sound/qcom,q6adm-routing.yaml# + unevaluatedProperties: false + description: Qualcomm DSP LPASS audio routing + +required: + - compatible + - routing + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@8 { + compatible = "qcom,q6adm"; + reg = <APR_SVC_ADM>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + routing { + compatible = "qcom,q6adm-routing"; + #sound-dai-cells = <0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt deleted file mode 100644 index 4916dd6a0896..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ /dev/null @@ -1,178 +0,0 @@ -Qualcomm Audio Front End (Q6AFE) binding - -AFE is one of the APR audio service on Q6DSP -Please refer to qcom,apr.txt for details of the common apr service bindings -used by all apr services. Must contain the following properties. - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>" - Or "qcom,q6afe" where the version number can be queried - 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 - -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>; - - dai@1 { - reg = <HDMI_RX>; - }; - - dai@24 { - reg = <PRIMARY_TDM_RX_0>; - 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>; - - }; - - dai@25 { - reg = <PRIMARY_TDM_TX_0>; - 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>; - }; - - dai@16 { - reg = <PRIMARY_MI2S_RX>; - qcom,sd-lines = <0 2>; - }; - - dai@17 { - reg = <PRIMARY_MI2S_TX>; - qcom,sd-lines = <1>; - }; - - dai@18 { - reg = <SECONDARY_MI2S_RX>; - qcom,sd-lines = <0 3>; - }; - - dai@19 { - reg = <SECONDARY_MI2S_TX>; - qcom,sd-lines = <1>; - }; - - dai@20 { - reg = <TERTIARY_MI2S_RX>; - qcom,sd-lines = <1 3>; - }; - - dai@21 { - reg = <TERTIARY_MI2S_TX>; - qcom,sd-lines = <0>; - }; - - dai@22 { - reg = <QUATERNARY_MI2S_RX>; - qcom,sd-lines = <0>; - }; - - dai@23 { - reg = <QUATERNARY_MI2S_TX>; - qcom,sd-lines = <1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.yaml b/Documentation/devicetree/bindings/sound/qcom,q6afe.yaml new file mode 100644 index 000000000000..297aa362aa54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio FrontEnd (Q6AFE) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6afe + + clock-controller: + $ref: /schemas/sound/qcom,q6dsp-lpass-clocks.yaml# + unevaluatedProperties: false + description: Qualcomm DSP LPASS clock controller + + dais: + type: object + $ref: /schemas/sound/qcom,q6dsp-lpass-ports.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + +required: + - compatible + - dais + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6afe.h> + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + clock-controller { + compatible = "qcom,q6afe-clocks"; + #clock-cells = <2>; + }; + + dais { + compatible = "qcom,q6afe-dais"; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@22 { + reg = <QUATERNARY_MI2S_RX>; + qcom,sd-lines = <0 1 2 3>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml new file mode 100644 index 000000000000..9e5b30d9c6e6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml @@ -0,0 +1,34 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6apm-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Process Manager Digital Audio Interfaces + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm APM DAIs in DSP + +properties: + compatible: + const: qcom,q6apm-dais + + iommus: + minItems: 1 + maxItems: 2 + +required: + - compatible + - iommus + +additionalProperties: false + +examples: + - | + dais { + compatible = "qcom,q6apm-dais"; + iommus = <&apps_smmu 0x1801 0x0>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml new file mode 100644 index 000000000000..894e653d37d7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6apm-lpass-dais.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm DSP LPASS (Low Power Audio SubSystem) Audio Ports + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - qcom,q6apm-lpass-dais + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + dais { + compatible = "qcom,q6apm-lpass-dais"; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm.yaml new file mode 100644 index 000000000000..ef1965aca254 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6apm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Process Manager (Q6APM) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6apm + + bedais: + type: object + $ref: /schemas/sound/qcom,q6apm-lpass-dais.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + + dais: + type: object + $ref: /schemas/sound/qcom,q6apm-dai.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - bedais + - dais + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,gpr.h> + + gpr { + #address-cells = <1>; + #size-cells = <0>; + + service@1 { + reg = <GPR_APM_MODULE_IID>; + compatible = "qcom,q6apm"; + #sound-dai-cells = <0>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + dais { + compatible = "qcom,q6apm-dais"; + iommus = <&apps_smmu 0x1801 0x0>; + }; + + bedais { + compatible = "qcom,q6apm-lpass-dais"; + #sound-dai-cells = <1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml b/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml new file mode 100644 index 000000000000..ce811942a9f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml @@ -0,0 +1,96 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6asm-dais.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Stream Manager (Q6ASM) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Q6ASM is one of the APR audio services on Q6DSP. Each of its subnodes + represent a dai with board specific configuration. + +properties: + compatible: + enum: + - qcom,q6asm-dais + + iommus: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + +patternProperties: + "^dai@[0-9]+$": + type: object + description: + Q6ASM Digital Audio Interface + + properties: + reg: + maxItems: 1 + + direction: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: | + The direction of the dai stream:: + - 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: + type: boolean + description: + Compress offload dai. + + dependencies: + is-compress-dai: [ direction ] + + required: + - reg + + additionalProperties: false + +required: + - compatible + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + dais { + compatible = "qcom,q6asm-dais"; + iommus = <&apps_smmu 0x1821 0x0>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@0 { + reg = <0>; + }; + + dai@1 { + reg = <1>; + }; + + dai@2 { + reg = <2>; + is-compress-dai; + direction = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt deleted file mode 100644 index 6b9a88d0ea3f..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ /dev/null @@ -1,69 +0,0 @@ -Qualcomm Audio Stream Manager (Q6ASM) binding - -Q6ASM is one of the APR audio service on Q6DSP. -Please refer to qcom,apr.txt for details of the common apr service bindings -used by the apr service device. - -- but must contain the following property: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>". - Or "qcom,q6asm" where the version number can be queried - from DSP. - example "qcom,q6asm-v2.0" - -= ASM DAIs (Digial Audio Interface) -"dais" subnode of the ASM node represents dai specific configuration - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6asm-dais". - -- #sound-dai-cells - Usage: required - Value type: <u32> - Definition: Must be 1 - -== ASM DAI is subnode of "dais" and represent a dai, it includes board specific -configuration of each dai. Must contain the following properties. - -- reg - Usage: required - Value type: <u32> - Definition: Must be dai id - -- direction: - Usage: Required for Compress offload dais - Value type: <u32> - Definition: Specifies the direction of the dai stream - 0 for both tx and rx - 1 for only tx (Capture/Encode) - 2 for only rx (Playback/Decode) - -- is-compress-dai: - Usage: Required for Compress offload dais - Value type: <boolean> - Definition: present for Compress offload dais - - -= EXAMPLE - -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>; - - dai@0 { - reg = <0>; - direction = <2>; - is-compress-dai; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6asm.yaml new file mode 100644 index 000000000000..cb49f9667cca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6asm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Stream Manager (Q6ASM) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6asm + + dais: + type: object + $ref: /schemas/sound/qcom,q6asm-dais.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + +required: + - compatible + - dais + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@7 { + compatible = "qcom,q6asm"; + reg = <APR_SVC_ASM>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + dais { + compatible = "qcom,q6asm-dais"; + iommus = <&apps_smmu 0x1821 0x0>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@0 { + reg = <0>; + }; + + dai@1 { + reg = <1>; + }; + + dai@2 { + reg = <2>; + is-compress-dai; + direction = <1>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt deleted file mode 100644 index 5cd4cc9b1fde..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ /dev/null @@ -1,21 +0,0 @@ -Qualcomm ADSP Core service binding - -Q6CORE is one of the APR audio service on Q6DSP. -Please refer to qcom,apr.txt for details of the common apr service bindings -used by the apr service device. - -- but must contain the following property: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>". - Or "qcom,q6core" where the version number can be queried - from DSP. - example "qcom,q6core-v2.0" - -= EXAMPLE -apr-service@3 { - compatible = "qcom,q6core"; - reg = <APR_SVC_ADSP_CORE>; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.yaml b/Documentation/devicetree/bindings/sound/qcom,q6core.yaml new file mode 100644 index 000000000000..e240712de9ca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6core.yaml @@ -0,0 +1,39 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6core.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Core (Q6Core) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6core + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@3 { + compatible = "qcom,q6core"; + reg = <APR_SVC_ADSP_CORE>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml new file mode 100644 index 000000000000..3552c44137ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-clocks.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm DSP LPASS Clock Controller + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP Clock Controller + +properties: + compatible: + enum: + - qcom,q6afe-clocks + - qcom,q6prm-lpass-clocks + + '#clock-cells': + const: 2 + description: + Clock Id is followed by clock coupling attributes. + 1 = for no coupled clock + 2 = for dividend of the coupled clock + 3 = for divisor of the coupled clock + 4 = for inverted and no couple clock + +required: + - compatible + - "#clock-cells" + +additionalProperties: false + +examples: + - | + clock-controller { + compatible = "qcom,q6afe-clocks"; + #clock-cells = <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml new file mode 100644 index 000000000000..08c618e7e428 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml @@ -0,0 +1,164 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-ports.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm DSP LPASS(Low Power Audio SubSystem) Audio Ports + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP LPASS Audio ports + +properties: + compatible: + enum: + - qcom,q6afe-dais + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +# Digital Audio Interfaces +patternProperties: + '^dai@[0-9]+$': + 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: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + Indicate how data is packed within the slot. For example, 32 slot + width in case of sample bit width is 24TDM Invert the sync. + 0 = MSB + 1 = LSB + + required: + - reg + + allOf: + - if: + properties: + reg: + contains: + # TDM DAI ID range from PRIMARY_TDM_RX_0 - QUINARY_TDM_TX_7 + items: + minimum: 24 + maximum: 103 + then: + required: + - qcom,tdm-sync-mode + - qcom,tdm-sync-src + - qcom,tdm-data-out + - qcom,tdm-invert-sync + - qcom,tdm-data-delay + - qcom,tdm-data-align + + - if: + properties: + reg: + contains: + # MI2S DAI ID range PRIMARY_MI2S_RX - QUATERNARY_MI2S_TX and + # QUINARY_MI2S_RX - QUINARY_MI2S_TX + items: + oneOf: + - minimum: 16 + maximum: 23 + - minimum: 127 + maximum: 128 + then: + required: + - qcom,sd-lines + + additionalProperties: false + +required: + - compatible + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6dsp-lpass-ports.h> + + dais { + compatible = "qcom,q6afe-dais"; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@22 { + reg = <QUATERNARY_MI2S_RX>; + qcom,sd-lines = <0 1 2 3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6prm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6prm.yaml new file mode 100644 index 000000000000..f6dbb1267bfe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6prm.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6prm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Proxy Resource Manager (Q6PRM) + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6prm + + clock-controller: + $ref: /schemas/sound/qcom,q6dsp-lpass-clocks.yaml# + unevaluatedProperties: false + description: Qualcomm DSP LPASS clock controller + +required: + - compatible + - clock-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,gpr.h> + + gpr { + #address-cells = <1>; + #size-cells = <0>; + + service@2 { + reg = <GPR_PRM_MODULE_IID>; + compatible = "qcom,q6prm"; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + clock-controller { + compatible = "qcom,q6prm-lpass-clocks"; + #clock-cells = <2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6usb.yaml b/Documentation/devicetree/bindings/sound/qcom,q6usb.yaml new file mode 100644 index 000000000000..37161d2aa96e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6usb.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6usb.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm ASoC DPCM USB backend DAI + +maintainers: + - Wesley Cheng <quic_wcheng@quicinc.com> + +description: + The USB port is a supported AFE path on the Q6 DSP. This ASoC DPCM + backend DAI will communicate the required settings to initialize the + XHCI host controller properly for enabling the offloaded audio stream. + Parameters defined under this node will carry settings, which will be + passed along during the QMI stream enable request and configuration of + the XHCI host controller. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - qcom,q6usb + + iommus: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + qcom,usb-audio-intr-idx: + description: + Desired XHCI interrupter number to use. Depending on the audio DSP + on the platform, it will operate on a specific XHCI interrupter. + $ref: /schemas/types.yaml#/definitions/uint16 + maximum: 8 + +required: + - compatible + - "#sound-dai-cells" + - qcom,usb-audio-intr-idx + +additionalProperties: false + +examples: + - | + dais { + compatible = "qcom,q6usb"; + #sound-dai-cells = <1>; + iommus = <&apps_smmu 0x180f 0x0>; + qcom,usb-audio-intr-idx = /bits/ 16 <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt deleted file mode 100644 index ca8c89e88bfa..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt +++ /dev/null @@ -1,83 +0,0 @@ -* Qualcomm Technologies Inc. SDM845 ASoC sound card driver - -This binding describes the SDM845 sound card, which uses qdsp for audio. - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be one of this - "qcom,sdm845-sndcard" - "qcom,db845c-sndcard" - "lenovo,yoga-c630-sndcard" - -- audio-routing: - Usage: Optional - Value type: <stringlist> - Definition: 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. - -- model: - Usage: required - Value type: <stringlist> - Definition: The user-visible name of this sound card. - -= dailinks -Each subnode of sndcard represents either a dailink, and subnodes of each -dailinks would be cpu/codec/platform dais. - -- link-name: - Usage: required - Value type: <string> - Definition: User friendly name for dai link - -= CPU, PLATFORM, CODEC dais subnodes -- cpu: - Usage: required - Value type: <subnode> - Definition: cpu dai sub-node - -- codec: - Usage: required - Value type: <subnode> - Definition: codec dai sub-node - -- platform: - Usage: Optional - Value type: <subnode> - Definition: platform dai sub-node - -- sound-dai: - Usage: required - Value type: <phandle> - Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. - -Example: - -audio { - compatible = "qcom,sdm845-sndcard"; - model = "sdm845-snd-card"; - pinctrl-names = "default", "sleep"; - pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>; - pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>; - - mm1-dai-link { - link-name = "MultiMedia1"; - cpu { - sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; - }; - }; - - pri-mi2s-dai-link { - link-name = "PRI MI2S Playback"; - cpu { - sound-dai = <&q6afedai PRIMARY_MI2S_RX>; - }; - - platform { - sound-dai = <&q6routing>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml new file mode 100644 index 000000000000..2ab6871e89e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -0,0 +1,325 @@ +# 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: + oneOf: + - items: + - enum: + - lenovo,yoga-c630-sndcard + - qcom,db845c-sndcard + - const: qcom,sdm845-sndcard + - items: + - enum: + - qcom,sm8550-sndcard + - qcom,sm8650-sndcard + - const: qcom,sm8450-sndcard + - enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + - qcom,qrb5165-rb5-sndcard + - qcom,sc7180-qdsp6-sndcard + - qcom,sc8280xp-sndcard + - qcom,sdm845-sndcard + - qcom,sm8250-sndcard + - qcom,sm8450-sndcard + - qcom,x1e80100-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 + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + 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,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt deleted file mode 100644 index 5d6ea66a863f..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt +++ /dev/null @@ -1,123 +0,0 @@ -QCOM WCD9335 Codec - -Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports -Qualcomm Technologies, Inc. (QTI) multimedia solutions, including -the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built -Soundwire controller, interrupt mux. It supports both I2S/I2C and -SLIMbus audio interfaces. - -Required properties with SLIMbus Interface: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: For SLIMbus interface it should be "slimMID,PID", - textual representation of Manufacturer ID, Product Code, - shall be in lower case hexadecimal with leading zeroes - suppressed. Refer to slimbus/bus.txt for details. - Should be: - "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus. - -- reg - Usage: required - Value type: <u32 u32> - Definition: Should be ('Device index', 'Instance ID') - -- interrupts - Usage: required - Value type: <prop-encoded-array> - Definition: Interrupts via WCD INTR1 and INTR2 pins - -- interrupt-names: - Usage: required - Value type: <String array> - Definition: Interrupt names of WCD INTR1 and INTR2 - Should be: "intr1", "intr2" - -- reset-gpios: - Usage: required - Value type: <String Array> - Definition: Reset gpio line - -- slim-ifc-dev: - Usage: required - Value type: <phandle> - Definition: SLIM interface device - -- clocks: - Usage: required - Value type: <prop-encoded-array> - Definition: See clock-bindings.txt section "consumers". List of - three clock specifiers for mclk, mclk2 and slimbus clock. - -- clock-names: - Usage: required - Value type: <string> - Definition: Must contain "mclk", "mclk2" and "slimbus" strings. - -- vdd-buck-supply: - Usage: required - Value type: <phandle> - Definition: Should contain a reference to the 1.8V buck supply - -- vdd-buck-sido-supply: - Usage: required - Value type: <phandle> - Definition: Should contain a reference to the 1.8V SIDO buck supply - -- vdd-rx-supply: - Usage: required - Value type: <phandle> - Definition: Should contain a reference to the 1.8V rx supply - -- vdd-tx-supply: - Usage: required - Value type: <phandle> - Definition: Should contain a reference to the 1.8V tx supply - -- vdd-vbat-supply: - Usage: Optional - Value type: <phandle> - Definition: Should contain a reference to the vbat supply - -- vdd-micbias-supply: - Usage: required - Value type: <phandle> - Definition: Should contain a reference to the micbias supply - -- vdd-io-supply: - Usage: required - Value type: <phandle> - Definition: Should contain a reference to the 1.8V io supply - -- interrupt-controller: - Usage: required - Definition: Indicating that this is a interrupt controller - -- #interrupt-cells: - Usage: required - Value type: <int> - Definition: should be 1 - -#sound-dai-cells - Usage: required - Value type: <u32> - Definition: Must be 1 - -audio-codec@1{ - compatible = "slim217,1a0"; - reg = <1 0>; - interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>; - interrupt-names = "intr2" - reset-gpios = <&msmgpio 64 0>; - slim-ifc-dev = <&wc9335_ifd>; - clock-names = "mclk", "native"; - clocks = <&rpmcc RPM_SMD_DIV_CLK1>, - <&rpmcc RPM_SMD_BB_CLK1>; - vdd-buck-supply = <&pm8994_s4>; - vdd-rx-supply = <&pm8994_s4>; - vdd-buck-sido-supply = <&pm8994_s4>; - vdd-tx-supply = <&pm8994_s4>; - vdd-io-supply = <&pm8994_s4>; - #sound-dai-cells = <1>; -} diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd9335.yaml new file mode 100644 index 000000000000..34f8fe4da9d4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.yaml @@ -0,0 +1,156 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd9335.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WCD9335 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC with in-built + Soundwire controller and interrupt mux. It supports both I2S/I2C and SLIMbus + audio interfaces. + +properties: + compatible: + const: slim217,1a0 + + reg: + maxItems: 1 + + clocks: + maxItems: 2 + + clock-names: + items: + - const: mclk + - const: slimbus + + interrupts: + maxItems: 2 + + interrupt-names: + items: + - const: intr1 + - const: intr2 + + interrupt-controller: true + + '#interrupt-cells': + const: 1 + + reset-gpios: + maxItems: 1 + + slim-ifc-dev: + description: SLIM IFC device interface + $ref: /schemas/types.yaml#/definitions/phandle + + '#sound-dai-cells': + const: 1 + + vdd-buck-supply: + description: 1.8V buck supply + + vdd-buck-sido-supply: + description: 1.8V SIDO buck supply + + vdd-io-supply: + description: 1.8V I/O supply + + vdd-micbias-supply: + description: micbias supply + + vdd-rx-supply: + description: 1.8V rx supply + + vdd-tx-supply: + description: 1.8V tx supply + + vdd-vbat-supply: + description: vbat supply + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + required: + - slim-ifc-dev + then: + required: + - clocks + - clock-names + - interrupts + - interrupt-names + - interrupt-controller + - '#interrupt-cells' + - reset-gpios + - slim-ifc-dev + - '#sound-dai-cells' + - vdd-buck-supply + - vdd-buck-sido-supply + - vdd-io-supply + - vdd-rx-supply + - vdd-tx-supply + else: + properties: + clocks: false + clock-names: false + interrupts: false + interrupt-names: false + interrupt-controller: false + '#interrupt-cells': false + reset-gpios: false + slim-ifc-dev: false + '#sound-dai-cells': false + vdd-buck-supply: false + vdd-buck-sido-supply: false + vdd-io-supply: false + vdd-micbias-supply: false + vdd-rx-supply: false + vdd-tx-supply: false + vdd-vbat-supply: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/qcom,rpmcc.h> + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + tasha_ifd: codec@0,0 { + compatible = "slim217,1a0"; + reg = <0 0>; + }; + + codec@1,0 { + compatible = "slim217,1a0"; + reg = <1 0>; + + clock-names = "mclk", "slimbus"; + clocks = <&div1_mclk>, <&rpmcc RPM_SMD_BB_CLK1>; + + interrupt-parent = <&tlmm>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH>, + <53 IRQ_TYPE_LEVEL_HIGH>; + interrupt-names = "intr1", "intr2"; + interrupt-controller; + #interrupt-cells = <1>; + + reset-gpios = <&tlmm 64 GPIO_ACTIVE_LOW>; + slim-ifc-dev = <&tasha_ifd>; + #sound-dai-cells = <1>; + + vdd-buck-supply = <&vreg_s4a_1p8>; + vdd-buck-sido-supply = <&vreg_s4a_1p8>; + vdd-tx-supply = <&vreg_s4a_1p8>; + vdd-rx-supply = <&vreg_s4a_1p8>; + vdd-io-supply = <&vreg_s4a_1p8>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index e8f716b5f875..beb0ff0245b0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -4,7 +4,7 @@ $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 +title: Qualcomm WCD9340/WCD9341 Audio Codec maintainers: - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> @@ -28,7 +28,9 @@ properties: description: GPIO spec for reset line to use maxItems: 1 - slim-ifc-dev: true + slim-ifc-dev: + description: IFC device interface + $ref: /schemas/types.yaml#/definitions/phandle clocks: maxItems: 1 @@ -77,6 +79,31 @@ properties: 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 @@ -102,11 +129,12 @@ properties: gpio@42: type: object - $ref: ../gpio/qcom,wcd934x-gpio.yaml# + $ref: /schemas/gpio/qcom,wcd934x-gpio.yaml# patternProperties: "^.*@[0-9a-f]+$": type: object + additionalProperties: true description: | WCD934x subnode for each slave devices. Bindings of each subnodes depends on the specific driver providing the functionality and @@ -122,21 +150,50 @@ patternProperties: 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" + +allOf: + - $ref: dai-common.yaml# + - if: + required: + - slim-ifc-dev + then: + required: + - reset-gpios + - slim-ifc-dev + - 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" + oneOf: + - required: + - interrupts-extended + - required: + - interrupts + else: + properties: + reset-gpios: false + slim-ifc-dev: false + interrupts: false + interrupt-controller: false + clock-frequency: false + clock-output-names: false + qcom,micbias1-microvolt: false + qcom,micbias2-microvolt: false + qcom,micbias3-microvolt: false + qcom,micbias4-microvolt: false + "#interrupt-cells": false + "#clock-cells": false + "#sound-dai-cells": false + "#address-cells": false + "#size-cells": false additionalProperties: false @@ -144,9 +201,9 @@ examples: - | codec@1,0{ compatible = "slim217,250"; - reg = <1 0>; + reg = <1 0>; reset-gpios = <&tlmm 64 0>; - slim-ifc-dev = <&wcd9340_ifd>; + slim-ifc-dev = <&wcd9340_ifd>; #sound-dai-cells = <1>; interrupt-parent = <&tlmm>; interrupts = <54 4>; @@ -159,6 +216,11 @@ examples: 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>; 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..7b31bf93f1a1 --- /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: 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. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml new file mode 100644 index 000000000000..cf6c3787adfe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: 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. + +allOf: + - $ref: dai-common.yaml# + - $ref: qcom,wcd93xx-common.yaml# + +properties: + compatible: + enum: + - qcom,wcd9380-codec + - qcom,wcd9385-codec + + us-euro-gpios: + description: GPIO spec for swapping gnd and mic segments + maxItems: 1 + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + codec { + compatible = "qcom,wcd9380-codec"; + reset-gpios = <&tlmm 32 0>; + #sound-dai-cells = <1>; + qcom,tx-device = <&wcd938x_tx>; + qcom,rx-device = <&wcd938x_rx>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + }; + + /* ... */ + + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml new file mode 100644 index 000000000000..67ed7701b5d6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd939x-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm SoundWire devices on WCD9390/WCD9395 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9390/WCD9395 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire devices. This bindings is for the devices. + +properties: + compatible: + const: sdw20217010e00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between device and host tx ports. + In the order of the device port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between device and host rx ports. + In the order of device port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010e00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5 6>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010e00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml new file mode 100644 index 000000000000..6e76f6a8634f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml @@ -0,0 +1,96 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd939x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WCD9380/WCD9385 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9390/WCD9395 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire devices. + The WCD9390/WCD9395 IC has a functionally separate USB-C Mux subsystem + accessible over an I2C interface. + The Audio Headphone and Microphone data path between the Codec and the USB-C Mux + subsystems are external to the IC, thus requiring DT port-endpoint graph description + to handle USB-C altmode & orientation switching for Audio Accessory Mode. + +allOf: + - $ref: dai-common.yaml# + - $ref: qcom,wcd93xx-common.yaml# + +properties: + compatible: + oneOf: + - const: qcom,wcd9390-codec + - items: + - const: qcom,wcd9395-codec + - const: qcom,wcd9390-codec + + mode-switch: + description: Flag the port as possible handler of altmode switching + type: boolean + + orientation-switch: + description: Flag the port as possible handler of orientation switching + type: boolean + + port: + $ref: /schemas/graph.yaml#/properties/port + description: + A port node to link the WCD939x Codec node to USB MUX subsystems for the + purpose of handling altmode muxing and orientation switching to detect and + enable Audio Accessory Mode. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + codec { + compatible = "qcom,wcd9390-codec"; + reset-gpios = <&tlmm 32 IRQ_TYPE_NONE>; + #sound-dai-cells = <1>; + qcom,tx-device = <&wcd939x_tx>; + qcom,rx-device = <&wcd939x_rx>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + }; + + /* ... */ + + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd939x_rx: codec@0,4 { + compatible = "sdw20217010e00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5 6>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010e00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml new file mode 100644 index 000000000000..f78ba148ad25 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml @@ -0,0 +1,95 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd93xx-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Common properties for Qualcomm WCD93xx Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + reset-gpios: + description: GPIO spec for reset line to use + 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: + - reset-gpios + - qcom,tx-device + - qcom,rx-device + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#sound-dai-cells" + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml index ea44d03e58ca..ac03672ebf6d 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml @@ -4,7 +4,7 @@ $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 +title: Qualcomm WSA8810/WSA8815 Class-D Smart Speaker Amplifier maintainers: - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> @@ -15,6 +15,9 @@ description: | Their primary operating mode uses a SoundWire digital audio interface. This binding is for SoundWire interface. +allOf: + - $ref: dai-common.yaml# + properties: compatible: const: sdw10217201000 @@ -39,7 +42,7 @@ required: - "#thermal-sensor-cells" - "#sound-dai-cells" -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml new file mode 100644 index 000000000000..8e462cdf0018 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa883x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8830/WSA8832/WSA8835 + smart speaker amplifier + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + WSA883X is the Qualcomm Aqstic smart speaker amplifier + Their primary operating mode uses a SoundWire digital audio + interface. This binding is for SoundWire interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw10217020200 + + reg: + maxItems: 1 + + powerdown-gpios: + description: GPIO spec for Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + vdd-supply: + description: VDD Supply for the Codec + + '#thermal-sensor-cells': + const: 0 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - vdd-supply + - powerdown-gpios + - "#thermal-sensor-cells" + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire@3250000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x3250000 0x2000>; + + speaker@0,1 { + compatible = "sdw10217020200"; + reg = <0 1>; + powerdown-gpios = <&tlmm 1 GPIO_ACTIVE_LOW>; + vdd-supply = <&vreg_s10b_1p8>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrLeft"; + }; + + speaker@0,2 { + compatible = "sdw10217020200"; + reg = <0 2>; + powerdown-gpios = <&tlmm 89 GPIO_ACTIVE_LOW>; + vdd-supply = <&vreg_s10b_1p8>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..22798d22d981 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + reset-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +oneOf: + - required: + - powerdown-gpios + - required: + - reset-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml b/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml new file mode 100644 index 000000000000..fb05988ff7ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,alc5632.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5632 audio CODEC + +description: | + Pins on the device (for linking into audio routes): + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + +maintainers: + - Leon Romanovsky <leon@leon.nu> + +properties: + compatible: + const: realtek,alc5632 + + reg: + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,alc5632"; + reg = <0x1a>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml new file mode 100644 index 000000000000..880196081a60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1015 Mono Class D Audio Amplifier + +maintainers: + - Jack Yu <jack.yu@realtek.com> + +properties: + compatible: + enum: + - realtek,rt1015 + + reg: + maxItems: 1 + + realtek,power-up-delay-ms: + description: Set a delay time for flush work to be completed, + this vlaue is adjustable depending on platform. + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml new file mode 100644 index 000000000000..7dac9e6f7f08 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015p.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt1015p codec + +maintainers: + - Tzung-Bi Shih <tzungbi@kernel.org> + +description: | + Rt1015p is a rt1015 variant which does not support I2C and + only supports S24, 48kHz, 64FS. + +properties: + compatible: + enum: + - realtek,rt1015p + - realtek,rt1019p + + sdb-gpios: + description: + GPIO used for shutdown control. + 0 means shut down; 1 means power on. + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + rt1015p: rt1015p { + compatible = "realtek,rt1015p"; + sdb-gpios = <&pio 175 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml new file mode 100644 index 000000000000..5287e9c9197e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1016.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Reaktek RT1016 Stereo Class D Audio Amplifier + +maintainers: + - oder_chiou@realtek.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt1016 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,rt1016"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml new file mode 100644 index 000000000000..248320804e5f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5616.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5616 ALSA SoC audio codec driver + +description: | + Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +maintainers: + - Bard Liao <bardliao@realtek.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5616 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml new file mode 100644 index 000000000000..ecfa7a576866 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -0,0 +1,150 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5682s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5682s codec + +maintainers: + - Derek Fang <derek.fang@realtek.com> + +description: | + Rt5682s(ALC5682I-VS) is a rt5682i variant which supports I2C only. + +allOf: + - $ref: dai-common.yaml# + +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. + + "#sound-dai-cells": + const: 1 + + AVDD-supply: + description: Regulator supplying analog power through the AVDD pin. + + MICVDD-supply: + description: Regulator supplying power for the microphone bias through the + MICVDD pin. + + DBVDD-supply: + description: Regulator supplying I/O power through the DBVDD pin. + + LDO1-IN-supply: + description: Regulator supplying power to the digital core and charge pump + through the LDO1_IN pin. + +unevaluatedProperties: false + +required: + - compatible + - reg + - AVDD-supply + - MICVDD-supply + - DBVDD-supply + - LDO1-IN-supply + +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"; + + AVDD-supply = <&avdd_reg>; + MICVDD-supply = <&micvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + LDO1-IN-supply = <&ldo1_in_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index 8a4406be387a..df91991699a7 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -9,6 +9,9 @@ title: Renesas FIFO-buffered Serial Interface (FSI) maintainers: - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +allOf: + - $ref: dai-common.yaml# + properties: $nodename: pattern: "^sound@.*" @@ -43,30 +46,19 @@ properties: '#sound-dai-cells': const: 1 - fsia,spdif-connection: +patternProperties: + "^fsi(a|b),spdif-connection$": $ref: /schemas/types.yaml#/definitions/flag description: FSI is connected by S/PDIF - fsia,stream-mode-support: + "^fsi(a|b),stream-mode-support$": $ref: /schemas/types.yaml#/definitions/flag description: FSI supports 16bit stream mode - fsia,use-internal-clock: + "^fsi(a|b),use-internal-clock$": $ref: /schemas/types.yaml#/definitions/flag description: FSI uses internal clock when master mode - fsib,spdif-connection: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,stream-mode-support: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,use-internal-clock: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - required: - compatible - reg @@ -75,7 +67,7 @@ required: - power-domains - '#sound-dai-cells' -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml b/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml new file mode 100644 index 000000000000..a2b92dba5529 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,idt821034.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas IDT821034 codec device + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The IDT821034 codec is a four channel PCM codec with onchip filters and + programmable gain setting. + + The time-slots used by the codec must be set and so, the properties + 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and + 'dai-tdm-slot-rx-mask' must be present in the ALSA sound card node for + sub-nodes that involve the codec. The codec uses one 8bit time-slot per + channel. + 'dai-tdm-tdm-slot-with' must be set to 8. + + The IDT821034 codec also supports 5 gpios (SLIC signals) per channel. + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + - $ref: dai-common.yaml# + +properties: + compatible: + const: renesas,idt821034 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 8192000 + + spi-cpha: true + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - spi-cpha + - '#sound-dai-cells' + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + compatible = "renesas,idt821034"; + reg = <0>; + spi-max-frequency = <8192000>; + spi-cpha; + #sound-dai-cells = <0>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 1596f0d1e2fe..dfd768b1ad7d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -94,7 +94,7 @@ see "Example: simple sound card for Asynchronous mode" [xx]ch [yy]ch ------> [CTU] --------> -CTU can convert [xx]ch to [yy]ch, or exchange outputed channel. +CTU can convert [xx]ch to [yy]ch, or exchange outputted channel. CTU conversion needs matrix settings. For more detail information, see below @@ -253,522 +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-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-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "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..0d7a6b576d88 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,538 @@ +# 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 + +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 + - const: 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 + - const: 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 + - const: renesas,rcar_sound-gen3 + # for Gen4 SoC + - items: + - const: renesas,rcar_sound-r8a779g0 # R-Car V4H + - const: renesas,rcar_sound-gen4 + # for Generic + - 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 + This is used on simple-audio-card + 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] + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + 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. + # details are defined below + + # ports is below + port: + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: audio-graph-port.yaml#/definitions/endpoint-base + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + unevaluatedProperties: false + + rcar_sound,dvc: + description: DVC subnode. + type: object + patternProperties: + "^dvc-[0-1]$": + type: object + additionalProperties: false + + 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]$": + type: object + additionalProperties: false + additionalProperties: false + + rcar_sound,ctu: + description: CTU subnode. + type: object + patternProperties: + "^ctu-[0-7]$": + type: object + additionalProperties: false + additionalProperties: false + + rcar_sound,src: + description: SRC subnode. + type: object + patternProperties: + "^src-[0-9]$": + type: object + additionalProperties: false + + properties: + interrupts: + maxItems: 1 + dmas: + maxItems: 2 + dma-names: + allOf: + - items: + enum: + - tx + - rx + additionalProperties: false + + rcar_sound,ssiu: + description: SSIU subnode. + type: object + patternProperties: + "^ssiu-[0-9]+$": + type: object + additionalProperties: false + + 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 + additionalProperties: false + + 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 + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + additionalProperties: false + +patternProperties: + # For DAI base + 'rcar_sound,dai(@[0-9a-f]+)?$': + description: DAI subnode. + type: object + patternProperties: + "^dai([0-9]+)?$": + type: object + additionalProperties: false + + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + + 'ports(@[0-9a-f]+)?$': + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + '^port(@[0-9a-f]+)?$': + $ref: "#/properties/port" + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + # -------------------- + # reg/reg-names + # -------------------- + # for Gen1 + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen1 + then: + properties: + reg: + maxItems: 3 + reg-names: + items: + enum: + - scu + - ssi + - adg + # for Gen2/Gen3 + - if: + properties: + compatible: + contains: + enum: + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + then: + properties: + reg: + minItems: 5 + reg-names: + items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + # for Gen4 + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + properties: + reg: + maxItems: 4 + reg-names: + items: + enum: + - adg + - ssiu + - ssi + - sdmc + + # -------------------- + # clock-names + # -------------------- + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + properties: + clock-names: + maxItems: 3 + items: + enum: + - ssi.0 + - ssiu.0 + - clkin + else: + properties: + clock-names: + minItems: 1 + maxItems: 31 + items: + oneOf: + - const: ssi-all + - pattern: '^ssi\.[0-9]$' + - pattern: '^src\.[0-9]$' + - pattern: '^mix\.[0-1]$' + - pattern: '^ctu\.[0-1]$' + - pattern: '^dvc\.[0-1]$' + - pattern: '^clk_(a|b|c|i)$' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/r8a7790-cpg-mssr.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/power/r8a7790-sysc.h> + 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 0x280>, /* SSI */ + <0xec740000 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&cpg CPG_MOD 1005>, /* SSI-ALL */ + <&cpg CPG_MOD 1006>, <&cpg CPG_MOD 1007>, /* SSI9, SSI8 */ + <&cpg CPG_MOD 1008>, <&cpg CPG_MOD 1009>, /* SSI7, SSI6 */ + <&cpg CPG_MOD 1010>, <&cpg CPG_MOD 1011>, /* SSI5, SSI4 */ + <&cpg CPG_MOD 1012>, <&cpg CPG_MOD 1013>, /* SSI3, SSI2 */ + <&cpg CPG_MOD 1014>, <&cpg CPG_MOD 1015>, /* SSI1, SSI0 */ + <&cpg CPG_MOD 1022>, <&cpg CPG_MOD 1023>, /* SRC9, SRC8 */ + <&cpg CPG_MOD 1024>, <&cpg CPG_MOD 1025>, /* SRC7, SRC6 */ + <&cpg CPG_MOD 1026>, <&cpg CPG_MOD 1027>, /* SRC5, SRC4 */ + <&cpg CPG_MOD 1028>, <&cpg CPG_MOD 1029>, /* SRC3, SRC2 */ + <&cpg CPG_MOD 1030>, <&cpg CPG_MOD 1031>, /* SRC1, SRC0 */ + <&cpg CPG_MOD 1020>, <&cpg CPG_MOD 1021>, /* MIX1, MIX0 */ + <&cpg CPG_MOD 1020>, <&cpg CPG_MOD 1021>, /* CTU1, CTU0 */ + <&cpg CPG_MOD 1019>, <&cpg CPG_MOD 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"; + + power-domains = <&sysc R8A7790_PD_ALWAYS_ON>; + + resets = <&cpg 1005>, + <&cpg 1006>, <&cpg 1007>, <&cpg 1008>, <&cpg 1009>, + <&cpg 1010>, <&cpg 1011>, <&cpg 1012>, <&cpg 1013>, + <&cpg 1014>, <&cpg 1015>; + reset-names = "ssi-all", + "ssi.9", "ssi.8", "ssi.7", "ssi.6", + "ssi.5", "ssi.4", "ssi.3", "ssi.2", + "ssi.1", "ssi.0"; + + 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 = <GIC_SPI 353 IRQ_TYPE_LEVEL_HIGH>; + 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 = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <GIC_SPI 371 IRQ_TYPE_LEVEL_HIGH>; + 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..8b9695f5decc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -0,0 +1,126 @@ +# 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> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + items: + - enum: + - renesas,r9a07g043-ssi # RZ/G2UL and RZ/Five + - renesas,r9a07g044-ssi # RZ/G2{L,LC} + - renesas,r9a07g054-ssi # RZ/V2L + - const: renesas,rz-ssi + + reg: + maxItems: 1 + + interrupts: + minItems: 2 + maxItems: 3 + + interrupt-names: + oneOf: + - items: + - const: int_req + - const: dma_rx + - const: dma_tx + - items: + - const: int_req + - 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' + +unevaluatedProperties: 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>; + interrupt-names = "int_req", "dma_rx", "dma_tx"; + 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..a1242e8e0687 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml @@ -0,0 +1,62 @@ +# 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. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - richtek,rt9120 + + reg: + description: I2C device address + maxItems: 1 + + pwdnn-gpios: + description: GPIO used for power down, low active + maxItems: 1 + + dvdd-supply: + description: | + Supply for the default on DVDD power, voltage domain must be 3P3V or 1P8V + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - dvdd-supply + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt9120@1a { + compatible = "richtek,rt9120"; + reg = <0x1a>; + pwdnn-gpios = <&gpio26 2 0>; + dvdd-supply = <&vdd_io_reg>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rtq9128.yaml b/Documentation/devicetree/bindings/sound/richtek,rtq9128.yaml new file mode 100644 index 000000000000..d54686a19ab7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rtq9128.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/richtek,rtq9128.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Richtek RTQ9128 Automative Audio Power Amplifier + +maintainers: + - ChiYuan Huang <cy_huang@richtek.com> + +description: + The RTQ9128 is a ultra-low output noise, high-efficiency, four-channel + class-D audio power amplifier and delivering 4x75W into 4OHm at 10% + THD+N from a 25V supply in automotive applications. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - richtek,rtq9128 + + reg: + maxItems: 1 + + enable-gpios: + maxItems: 1 + + richtek,tdm-input-data2-select: + type: boolean + description: + By default, if TDM mode is used, TDM data input will select 'DATA1' pin + as the data source. This option will configure TDM data input source from + 'DATA1' to 'DATA2' pin. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + speaker@1a { + compatible = "richtek,rtq9128"; + reg = <0x1a>; + enable-gpios = <&gpio 26 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml new file mode 100644 index 000000000000..7bb6c5dff786 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -0,0 +1,192 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - rockchip,px30-i2s-tdm + - rockchip,rk1808-i2s-tdm + - rockchip,rk3308-i2s-tdm + - rockchip,rk3568-i2s-tdm + - rockchip,rk3588-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 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + power-domains: + maxItems: 1 + + 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 + - "#sound-dai-cells" + +unevaluatedProperties: 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..ff9e400494f3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml @@ -0,0 +1,123 @@ +# 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> + +allOf: + - $ref: dai-common.yaml# + +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" + +unevaluatedProperties: 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,rk3288-hdmi-analog.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt index e5430d1d34e4..73577ac1b89c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt @@ -12,7 +12,7 @@ Required properties: source. For this driver the first string should always be "Analog". -Optionnal properties: +Optional properties: - rockchip,hp-en-gpios = The phandle of the GPIO that power up/down the headphone (when the analog output is an headphone). - rockchip,hp-det-gpios = The phandle of the GPIO that detects the headphone 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 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +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. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -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"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; 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..5cdb8bcc687b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.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/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko@sntech.de> +allOf: + - $ref: dai-common.yaml# + + +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" + +unevaluatedProperties: 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.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml index acb2b888dbfc..fcb01abffa97 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -13,22 +13,29 @@ description: maintainers: - Heiko Stuebner <heiko@sntech.de> +allOf: + - $ref: dai-common.yaml# + properties: compatible: oneOf: - const: rockchip,rk3066-i2s - items: - enum: - - rockchip,px30-i2s - - rockchip,rk3036-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,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,rk3588-i2s + - rockchip,rv1126-i2s - const: rockchip,rk3066-i2s reg: @@ -55,12 +62,31 @@ properties: oneOf: - const: rx - items: - - const: tx - - const: rx + - const: tx + - const: rx + + pinctrl-names: + oneOf: + - const: default + - items: + - const: bclk_on + - const: bclk_off power-domains: maxItems: 1 + reset-names: + items: + - const: reset-m + - const: reset-h + + resets: + maxItems: 2 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + rockchip,capture-channels: $ref: /schemas/types.yaml#/definitions/uint32 default: 2 @@ -93,7 +119,7 @@ required: - dma-names - "#sound-dai-cells" -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index c467152656f7..c3c989ef2a2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -23,10 +23,13 @@ properties: - 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,rk3128-spdif + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - rockchip,rk3308-spdif - const: rockchip,rk3066-spdif reg: @@ -73,17 +76,18 @@ required: - dma-names - "#sound-dai-cells" -if: - properties: - compatible: - contains: - const: rockchip,rk3288-spdif - -then: - required: - - rockchip,grf - -additionalProperties: false +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + then: + required: + - rockchip,grf + +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..5abcf92bc484 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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> + +allOf: + - $ref: dai-common.yaml# + +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" + +unevaluatedProperties: 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/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt deleted file mode 100644 index fcfd02d8d32f..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ /dev/null @@ -1,17 +0,0 @@ -RT1015 Mono Class D Audio Amplifier - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt1015". - -- reg : The I2C address of the device. - - -Example: - -rt1015: codec@28 { - compatible = "realtek,rt1015"; - reg = <0x28>; -}; diff --git a/Documentation/devicetree/bindings/sound/rt1016.txt b/Documentation/devicetree/bindings/sound/rt1016.txt deleted file mode 100644 index 2310f8ff259b..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1016.txt +++ /dev/null @@ -1,17 +0,0 @@ -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/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt deleted file mode 100644 index 540a4bf252e4..000000000000 --- a/Documentation/devicetree/bindings/sound/rt5616.txt +++ /dev/null @@ -1,32 +0,0 @@ -RT5616 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt5616". - -- reg : The I2C address of the device. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC. - -- clock-names: Should be "mclk". - -Pins on the device (for linking into audio routes) for RT5616: - - * IN1P - * IN2P - * IN2N - * LOUTL - * LOUTR - * HPOL - * HPOR - -Example: - -rt5616: codec@1b { - compatible = "realtek,rt5616"; - reg = <0x1b>; -}; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..0c398581d52b 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -20,6 +20,9 @@ Optional properties: - realtek,in3-differential Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. +- realtek,lout-differential + Boolean. Indicate LOUT output is differential, rather than stereo. + - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. - realtek,dmic1-data-pin @@ -88,7 +91,7 @@ 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/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..8f3f62c0226a 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 dai-common.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/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 2a55e9133408..24a6dab28f25 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -28,7 +28,7 @@ Optional properties: If the value is 0, it means the impedance sensing is not supported. - "realtek,impedance_sensing_table" The matrix rows of the impedance sensing table are consisted by impedance - minimum, impedance maximun, volume, DC offset w/o and w/ mic of each L and + minimum, impedance maximum, volume, DC offset w/o and w/ mic of each L and R channel accordingly. Example is shown as following. < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0> 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 ade1ece8b45f..5e1d08de18a5 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -8,6 +8,21 @@ Required properties: - reg : The I2C address of the device. +- AVDD-supply: phandle to the regulator supplying analog power through the + AVDD pin + +- MICVDD-supply: phandle to the regulator supplying power for the microphone + bias through the MICVDD pin. Either MICVDD or VBAT should be present. + +- VBAT-supply: phandle to the regulator supplying battery power through the + VBAT pin. Either MICVDD or VBAT should be present. + +- DBVDD-supply: phandle to the regulator supplying I/O power through the DBVDD + pin. + +- LDO1-IN-supply: phandle to the regulator supplying power to the digital core + and charge pump through the LDO1_IN pin. + Optional properties: - interrupts : The CODEC's interrupt output. @@ -44,6 +59,10 @@ Optional properties: - 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 '<1>'. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 @@ -58,7 +77,7 @@ 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>; @@ -71,4 +90,9 @@ rt5682 { clocks = <&osc>; clock-names = "mclk"; + + AVDD-supply = <&avdd_reg>; + MICVDD-supply = <&micvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + LDO1-IN-supply = <&ldo1_in_reg>; }; 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..5ea0819a261a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.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/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> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 + + cpu: + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 2 + maxItems: 2 + description: | + phandles to the I2S controller and bluetooth codec, + in that order + required: + - sound-dai + + codec: + additionalProperties: false + 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 + deprecated: true + 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 + Deprecated, use audio-routing. + + 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 + - cpu + - codec + - audio-routing + - extcon + - main-micbias-supply + - headset-micbias-supply + - earpath-sel-gpios + - headset-detect-gpios + - headset-key-gpios + +unevaluatedProperties: 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>; + + 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. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - samsung,audio-codec + - samsung,audio-cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,arndale-rt5631"; + samsung,audio-cpu = <&i2s0>; + samsung,audio-codec = <&rt5631>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 000000000000..6ec80f529d84 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,112 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: samsung,midas-audio + + cpu: + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + description: phandle to the I2S controller + required: + - sound-dai + + codec: + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + description: phandle to the WM1811 CODEC + required: + - sound-dai + + samsung,audio-routing: + deprecated: true + $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. + Deprecated, use audio-routing. + + 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 + - cpu + - codec + - audio-routing + - mic-bias-supply + - submic-bias-supply + +unevaluatedProperties: 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>; + + 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.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index 8ff2d39e7d17..b77284e3e26a 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -10,6 +10,9 @@ maintainers: - Krzysztof Kozlowski <krzk@kernel.org> - Sylwester Nawrocki <s.nawrocki@samsung.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: oneOf: @@ -24,29 +27,31 @@ properties: - 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 + additionalProperties: false properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array description: phandles to the I2S controllers codec: type: object + additionalProperties: false properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array - description: | - 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. + minItems: 1 + items: + - description: phandle of the HDMI IP block node + - description: phandle of the MAX98090 CODEC samsung,audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array + deprecated: true description: | List of the connections between audio components; each entry is a pair of strings, the first being the @@ -56,6 +61,7 @@ properties: For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" For Odroid U3, XU3: "Headphone Jack", "Speakers" For Odroid XU4: no entries + Deprecated, use audio-routing. samsung,audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array @@ -65,18 +71,17 @@ properties: required: - compatible - - model - cpu - codec -additionalProperties: false +unevaluatedProperties: false examples: - | sound { compatible = "hardkernel,odroid-xu3-audio"; model = "Odroid-XU3"; - samsung,audio-routing = + audio-routing = "Headphone Jack", "HPL", "Headphone Jack", "HPR", "IN1", "Mic Jack", @@ -90,4 +95,3 @@ examples: sound-dai = <&hdmi>, <&max98090>; }; }; - diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt deleted file mode 100644 index 4686646fb122..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt +++ /dev/null @@ -1,14 +0,0 @@ -Samsung SMDK audio complex - -Required properties: -- compatible : "samsung,smdk-wm8994" -- samsung,i2s-controller: The phandle of the Samsung I2S0 controller -- samsung,audio-codec: The phandle of the WM8994 audio codec -Example: - -sound { - compatible = "samsung,smdk-wm8994"; - - samsung,i2s-controller = <&i2s0>; - samsung,audio-codec = <&wm8994>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml new file mode 100644 index 000000000000..ac151d3c1d77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + $ref: /schemas/types.yaml#/definitions/phandle + +required: + - compatible + - samsung,audio-codec + - samsung,i2s-controller + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,smdk-wm8994"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&wm8994>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml new file mode 100644 index 000000000000..3d49aa4c9be2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + additionalProperties: false + 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 + additionalProperties: false + 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..cbc7ba37362a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2.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/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> + +allOf: + - $ref: sound-card-common.yaml# + +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: + - items: + - description: Phandle to the WM5110 audio codec. + - items: + - 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. + Deprecated, use audio-routing. + deprecated: true + $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. + +required: + - compatible + - audio-amplifier + - audio-codec + - audio-routing + - i2s-controller + - mic-bias-gpios + +unevaluatedProperties: 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"; + 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.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml index b2ad093d94df..f45f73b5056d 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -10,6 +10,9 @@ maintainers: - Krzysztof Kozlowski <krzk@kernel.org> - Sylwester Nawrocki <s.nawrocki@samsung.com> +allOf: + - $ref: dai-common.yaml# + properties: compatible: description: | @@ -34,12 +37,31 @@ properties: 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 + + tesla,fsd-i2s: for 8/16/24bit stereo channel I2S for playback and + capture, secondary FIFO using external DMA, s/w reset control, + internal mux for root clock source with all root clock sampling + frequencies supported by Exynos7 I2S and 7.1 channel TDM support + for playback and capture TDM (Time division multiplexing) to allow + transfer of multiple channel audio data on single data line. + oneOf: + - enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + - tesla,fsd-i2s + - items: + - enum: + - samsung,exynos5433-i2s + - const: samsung,exynos7-i2s + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 reg: maxItems: 1 @@ -92,17 +114,17 @@ properties: - const: i2s_cdclk2 description: Names of the CDCLK I2S output clocks. + interrupts: + maxItems: 1 + samsung,idma-addr: $ref: /schemas/types.yaml#/definitions/uint32 description: | Internal DMA register base address of the audio subsystem (used in secondary sound source). - pinctrl-0: - description: Should specify pin control groups used for this controller. - - pinctrl-names: - const: default + power-domains: + maxItems: 1 "#sound-dai-cells": const: 1 @@ -115,7 +137,7 @@ required: - clocks - clock-names -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/serial-midi.yaml b/Documentation/devicetree/bindings/sound/serial-midi.yaml new file mode 100644 index 000000000000..f6a807329a5a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/serial-midi.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause + +%YAML 1.2 +--- +$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 + results 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..1353c051488f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -0,0 +1,113 @@ +# 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> + +allOf: + - $ref: dai-common.yaml# + +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. + $ref: /schemas/types.yaml#/definitions/uint32 + 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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <®_3p3v>; + VDDIO-supply = <®_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt deleted file mode 100644 index b1b097cc9b68..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ /dev/null @@ -1,17 +0,0 @@ -Simple Amplifier Audio Driver - -Required properties: -- compatible : "dioo,dio2125" or "simple-audio-amplifier" - -Optional properties: -- enable-gpios : the gpio connected to the enable pin of the simple amplifier -- VCC-supply : power supply for the device, as covered - in Documentation/devicetree/bindings/regulator/regulator.txt - -Example: - -amp: analog-amplifier { - compatible = "simple-audio-amplifier"; - VCC-supply = <®ulator>; - enable-gpios = <&gpio GPIOH_3 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml new file mode 100644 index 000000000000..5db1f989d050 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-amplifier.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Amplifier + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - dioo,dio2125 + - simple-audio-amplifier + + enable-gpios: + maxItems: 1 + + VCC-supply: + description: > + power supply for the device + + sound-name-prefix: true + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/meson8-gpio.h> + + analog-amplifier { + compatible = "simple-audio-amplifier"; + VCC-supply = <®ulator>; + enable-gpios = <&gpio GPIOH_3 0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000000..9f319caf3db7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$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: dai-common.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.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index 8132d0c0f00a..59ac2d1d1ccf 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -4,7 +4,7 @@ $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 +title: Simple Audio Card Driver maintainers: - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> @@ -13,13 +13,11 @@ definitions: frame-master: description: Indicates dai-link frame master. - $ref: /schemas/types.yaml#/definitions/phandle-array - maxItems: 1 + $ref: /schemas/types.yaml#/definitions/phandle bitclock-master: description: Indicates dai-link bit clock master - $ref: /schemas/types.yaml#/definitions/phandle-array - maxItems: 1 + $ref: /schemas/types.yaml#/definitions/phandle frame-inversion: description: dai-link uses frame clock inversion @@ -50,6 +48,15 @@ definitions: 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. @@ -71,7 +78,7 @@ definitions: $ref: /schemas/types.yaml#/definitions/uint32 prefix: - description: "device name prefix" + description: device name prefix $ref: /schemas/types.yaml#/definitions/string label: @@ -136,9 +143,20 @@ definitions: $ref: "#/definitions/system-clock-frequency" system-clock-direction-out: $ref: "#/definitions/system-clock-direction-out" + system-clock-fixed: + $ref: "#/definitions/system-clock-fixed" required: - sound-dai + additional-devs: + type: object + description: + Additional devices used by the simple audio card. + patternProperties: + '^iio-aux(-.+)?$': + type: object + $ref: audio-iio-aux.yaml# + properties: compatible: contains: @@ -158,45 +176,49 @@ properties: description: User specified audio sound card name. $ref: /schemas/types.yaml#/definitions/string -# use patternProperties to avoid naming "xxx,yyy" issue -patternProperties: - "^simple-audio-card,widgets$": + simple-audio-card,widgets: $ref: "#/definitions/widgets" - "^simple-audio-card,routing$": + simple-audio-card,routing: $ref: "#/definitions/routing" - "^simple-audio-card,cpu(@[0-9a-f]+)?": - $ref: "#/definitions/dai" - "^simple-audio-card,codec(@[0-9a-f]+)?": - $ref: "#/definitions/dai" # common properties - "^simple-audio-card,frame-master$": + simple-audio-card,frame-master: $ref: "#/definitions/frame-master" - "^simple-audio-card,bitclock-master$": + simple-audio-card,bitclock-master: $ref: "#/definitions/bitclock-master" - "^simple-audio-card,frame-inversion$": + simple-audio-card,frame-inversion: $ref: "#/definitions/frame-inversion" - "^simple-audio-card,bitclock-inversion$": + simple-audio-card,bitclock-inversion: $ref: "#/definitions/bitclock-inversion" - "^simple-audio-card,format$": + simple-audio-card,format: $ref: "#/definitions/format" - "^simple-audio-card,mclk-fs$": + simple-audio-card,mclk-fs: $ref: "#/definitions/mclk-fs" - "^simple-audio-card,aux-devs$": + simple-audio-card,aux-devs: $ref: "#/definitions/aux-devs" - "^simple-audio-card,convert-rate$": + simple-audio-card,additional-devs: + $ref: "#/definitions/additional-devs" + simple-audio-card,convert-rate: $ref: "#/definitions/convert-rate" - "^simple-audio-card,convert-channels$": + simple-audio-card,convert-channels: $ref: "#/definitions/convert-channels" - "^simple-audio-card,prefix$": + simple-audio-card,prefix: $ref: "#/definitions/prefix" - "^simple-audio-card,pin-switches$": + simple-audio-card,pin-switches: $ref: "#/definitions/pin-switches" - "^simple-audio-card,hp-det-gpio$": + simple-audio-card,hp-det-gpio: maxItems: 1 - "^simple-audio-card,mic-det-gpio$": + simple-audio-card,mic-det-gpio: maxItems: 1 +patternProperties: + "^simple-audio-card,cpu(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,plat(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": description: | Container for dai-link level properties and the CPU and CODEC sub-nodes. @@ -206,6 +228,10 @@ patternProperties: reg: maxItems: 1 + "#address-cells": + const: 1 + "#size-cells": + const: 0 # common properties frame-master: $ref: "#/definitions/frame-master" @@ -235,9 +261,9 @@ patternProperties: maxItems: 1 patternProperties: - "^cpu(@[0-9a-f]+)?": + "^cpu(-[0-9]+)?$": $ref: "#/definitions/dai" - "^codec(@[0-9a-f]+)?": + "^codec(-[0-9]+)?$": $ref: "#/definitions/dai" additionalProperties: false @@ -247,9 +273,9 @@ required: additionalProperties: false examples: -#-------------------- +# -------------------- # single DAI link -#-------------------- +# -------------------- - | sound { compatible = "simple-audio-card"; @@ -276,9 +302,9 @@ examples: }; }; -#-------------------- +# -------------------- # Multi DAI links -#-------------------- +# -------------------- - | sound { compatible = "simple-audio-card"; @@ -319,10 +345,10 @@ examples: }; }; -#-------------------- +# -------------------- # route audio from IMX6 SSI2 through TLV320DAC3100 codec # through TPA6130A2 amplifier to headphones: -#-------------------- +# -------------------- - | sound { compatible = "simple-audio-card"; @@ -344,9 +370,51 @@ examples: }; }; -#-------------------- +# -------------------- +# route audio to/from a codec through an amplifier +# designed with a potentiometer driven by IIO: +# -------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,aux-devs = <&_in>, <&_out>; + simple-audio-card,routing = + "CODEC LEFTIN", "AMP_IN LEFT OUT", + "CODEC RIGHTIN", "AMP_IN RIGHT OUT", + "AMP_OUT LEFT IN", "CODEC LEFTOUT", + "AMP_OUT RIGHT IN", "CODEC RIGHTOUT"; + + simple-audio-card,additional-devs { + amp_out: iio-aux-out { + compatible = "audio-iio-aux"; + io-channels = <&pot_out 0>, <&pot_out 1>; + io-channel-names = "LEFT", "RIGHT"; + snd-control-invert-range = <1 1>; + sound-name-prefix = "AMP_OUT"; + }; + + amp_in: iio_aux-in { + compatible = "audio-iio-aux"; + io-channels = <&pot_in 0>, <&pot_in 1>; + io-channel-names = "LEFT", "RIGHT"; + sound-name-prefix = "AMP_IN"; + }; + }; + + simple-audio-card,cpu { + sound-dai = <&cpu>; + }; + + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +# -------------------- # Sampling Rate Conversion -#-------------------- +# -------------------- - | sound { compatible = "simple-audio-card"; @@ -372,12 +440,14 @@ examples: }; }; -#-------------------- +# -------------------- # 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"; @@ -391,10 +461,12 @@ examples: "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>; }; @@ -405,7 +477,7 @@ examples: }; }; -#-------------------- +# -------------------- # Multi DAI links with DPCM: # # CPU0 ------ ak4613 @@ -414,10 +486,12 @@ examples: # 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", @@ -426,6 +500,7 @@ examples: "pcm3168a Playback", "DAI4 Playback"; simple-audio-card,dai-link@0 { + reg = <0>; format = "left_j"; bitclock-master = <&sndcpu0>; frame-master = <&sndcpu0>; @@ -439,22 +514,23 @@ examples: }; simple-audio-card,dai-link@1 { + reg = <1>; format = "i2s"; bitclock-master = <&sndcpu1>; frame-master = <&sndcpu1>; convert-channels = <8>; /* TDM Split */ - sndcpu1: cpu@0 { + sndcpu1: cpu-0 { sound-dai = <&rcar_sound 1>; }; - cpu@1 { + cpu-1 { sound-dai = <&rcar_sound 2>; }; - cpu@2 { + cpu-2 { sound-dai = <&rcar_sound 3>; }; - cpu@3 { + cpu-3 { sound-dai = <&rcar_sound 4>; }; codec { @@ -466,6 +542,7 @@ examples: }; simple-audio-card,dai-link@2 { + reg = <2>; format = "i2s"; bitclock-master = <&sndcpu2>; frame-master = <&sndcpu2>; 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/snps,designware-i2s.yaml b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml new file mode 100644 index 000000000000..a48d040b0a4f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml @@ -0,0 +1,197 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/snps,designware-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: DesignWare I2S controller + +maintainers: + - Jose Abreu <joabreu@synopsys.com> + +properties: + compatible: + oneOf: + - items: + - const: canaan,k210-i2s + - const: snps,designware-i2s + - enum: + - snps,designware-i2s + - starfive,jh7110-i2stx0 + - starfive,jh7110-i2stx1 + - starfive,jh7110-i2srx + + reg: + maxItems: 1 + + interrupts: + description: | + The interrupt line number for the I2S controller. Add this + parameter if the I2S controller that you are using does not + support DMA. + maxItems: 1 + + clocks: + items: + - description: Sampling rate reference clock + - description: APB clock + - description: Audio master clock + - description: Inner audio master clock source + - description: External audio master clock source + - description: Bit clock + - description: Left/right channel clock + - description: External bit clock + - description: External left/right channel clock + minItems: 1 + + clock-names: + items: + - const: i2sclk + - const: apb + - const: mclk + - const: mclk_inner + - const: mclk_ext + - const: bclk + - const: lrck + - const: bclk_ext + - const: lrck_ext + minItems: 1 + + resets: + items: + - description: Optional controller resets + - description: controller reset of Sampling rate + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + minItems: 1 + + dma-names: + items: + - const: tx + - const: rx + minItems: 1 + + starfive,syscon: + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: phandle to System Register Controller sys_syscon node. + - description: I2S-rx enabled control offset of SYS_SYSCONSAIF__SYSCFG register. + - description: I2S-rx enabled control mask + description: + The phandle to System Register Controller syscon node and the I2S-rx(ADC) + enabled control offset and mask of SYS_SYSCONSAIF__SYSCFG register. + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: canaan,k210-i2s + then: + properties: + "#sound-dai-cells": + const: 1 + else: + properties: + "#sound-dai-cells": + const: 0 + - if: + properties: + compatible: + contains: + const: snps,designware-i2s + then: + properties: + clocks: + maxItems: 1 + clock-names: + maxItems: 1 + resets: + maxItems: 1 + else: + properties: + resets: + minItems: 2 + maxItems: 2 + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2stx0 + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + minItems: 5 + maxItems: 5 + required: + - resets + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2stx1 + then: + properties: + clocks: + minItems: 9 + maxItems: 9 + clock-names: + minItems: 9 + maxItems: 9 + required: + - resets + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2srx + then: + properties: + clocks: + minItems: 9 + maxItems: 9 + clock-names: + minItems: 9 + maxItems: 9 + required: + - resets + - starfive,syscon + else: + properties: + starfive,syscon: false + +required: + - compatible + - reg + - clocks + - clock-names + +oneOf: + - required: + - dmas + - dma-names + - required: + - interrupts + +unevaluatedProperties: false + +examples: + - | + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x7ff90000 0x1000>; + clocks = <&scpi_i2sclk 0>; + clock-names = "i2sclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000000..8600520d7c47 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,102 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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> + +allOf: + - $ref: dai-common.yaml# + +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 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +unevaluatedProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000000..985277648de1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels. + 0: line1 + 1: hp + 2: line2 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +unevaluatedProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml new file mode 100644 index 000000000000..721950f65748 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -0,0 +1,34 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-card-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Board Sound Card Common Properties + +maintainers: + - Mark Brown <broonie@kernel.org> + +properties: + 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. + + ignore-suspend-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of audio sound widgets which are marked ignoring system suspend. + Paths between these endpoints are still active over suspend of the main + application processor that the current operating system is running. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +required: + - model + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/sound-dai.yaml b/Documentation/devicetree/bindings/sound/sound-dai.yaml new file mode 100644 index 000000000000..ff9036e43c44 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-dai.yaml @@ -0,0 +1,20 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface consumer + +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/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt index 4b23e84b2e57..fbbcade2181d 100644 --- a/Documentation/devicetree/bindings/sound/sprd-pcm.txt +++ b/Documentation/devicetree/bindings/sound/sprd-pcm.txt @@ -1,4 +1,4 @@ -* Spreadtrum DMA platfrom bindings +* Spreadtrum DMA platform bindings Required properties: - compatible: Should be "sprd,pcm-platform". diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt deleted file mode 100644 index 59381a778c79..000000000000 --- a/Documentation/devicetree/bindings/sound/ssm2518.txt +++ /dev/null @@ -1,20 +0,0 @@ -SSM2518 audio amplifier - -This device supports I2C only. - -Required properties: - - compatible : Must be "adi,ssm2518" - - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) - or 0x35 (ADDR pin high) - -Optional properties: - - gpios : GPIO connected to the nSD pin. If the property is not present it is - assumed that the nSD pin is hardwired to always on. - -Example: - - ssm2518: ssm2518@34 { - compatible = "adi,ssm2518"; - reg = <0x34>; - gpios = <&gpio 5 0>; - }; 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.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml index f32410890589..b9111d375b93 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -7,12 +7,15 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 SPI/I2S Controller maintainers: - - Olivier Moysan <olivier.moysan@st.com> + - 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. +allOf: + - $ref: dai-common.yaml# + properties: compatible: enum: @@ -54,6 +57,14 @@ properties: resets: maxItems: 1 + "#clock-cells": + description: Configure the I2S device as MCLK clock provider. + const: 0 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - "#sound-dai-cells" @@ -64,7 +75,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | @@ -82,6 +93,13 @@ examples: dma-names = "rx", "tx"; pinctrl-names = "default"; pinctrl-0 = <&i2s2_pins_a>; + + /* assume audio-graph */ + port { + codec_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + }; + }; }; ... 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 c42b91e525fa..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/st,stm32-dma.yaml - - 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..59df8a832310 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.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/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 + additionalProperties: false + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unused 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 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + 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.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml index b7f7dc452231..bc48151b9adb 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -7,12 +7,15 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) maintainers: - - Olivier Moysan <olivier.moysan@st.com> + - 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. +allOf: + - $ref: dai-common.yaml# + properties: compatible: enum: @@ -57,7 +60,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml b/Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml new file mode 100644 index 000000000000..e2b4db6aa2fb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/starfive,jh7110-pwmdac.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: StarFive JH7110 PWM-DAC Controller + +description: + The PWM-DAC Controller uses PWM square wave generators plus RC filters to + form a DAC for audio play in StarFive JH7110 SoC. This audio play controller + supports 16 bit audio format, up to 48K sampling frequency, up to left and + right dual channels. + +maintainers: + - Hal Feng <hal.feng@starfivetech.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: starfive,jh7110-pwmdac + + reg: + maxItems: 1 + + clocks: + items: + - description: PWMDAC APB + - description: PWMDAC CORE + + clock-names: + items: + - const: apb + - const: core + + resets: + maxItems: 1 + description: PWMDAC APB + + dmas: + maxItems: 1 + description: TX DMA Channel + + dma-names: + const: tx + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - resets + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + pwmdac@100b0000 { + compatible = "starfive,jh7110-pwmdac"; + reg = <0x100b0000 0x1000>; + clocks = <&syscrg 157>, + <&syscrg 158>; + clock-names = "apb", "core"; + resets = <&syscrg 96>; + dmas = <&dma 22>; + dma-names = "tx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml b/Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml new file mode 100644 index 000000000000..abb373fbfa26 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/starfive,jh7110-tdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: StarFive JH7110 TDM Controller + +description: | + The TDM Controller is a Time Division Multiplexed audio interface + integrated in StarFive JH7110 SoC, allowing up to 8 channels of + audio over a serial interface. The TDM controller can operate both + in master and slave mode. + +maintainers: + - Walker Chen <walker.chen@starfivetech.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - starfive,jh7110-tdm + + reg: + maxItems: 1 + + clocks: + items: + - description: TDM AHB Clock + - description: TDM APB Clock + - description: TDM Internal Clock + - description: TDM Clock + - description: Inner MCLK + - description: TDM External Clock + + clock-names: + items: + - const: tdm_ahb + - const: tdm_apb + - const: tdm_internal + - const: tdm + - const: mclk_inner + - const: tdm_ext + + resets: + items: + - description: tdm ahb reset line + - description: tdm apb reset line + - description: tdm core reset line + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - resets + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + tdm@10090000 { + compatible = "starfive,jh7110-tdm"; + reg = <0x10090000 0x1000>; + clocks = <&syscrg 184>, + <&syscrg 185>, + <&syscrg 186>, + <&syscrg 187>, + <&syscrg 17>, + <&tdm_ext>; + clock-names = "tdm_ahb", "tdm_apb", + "tdm_internal", "tdm", + "mclk_inner", "tdm_ext"; + resets = <&syscrg 105>, + <&syscrg 107>, + <&syscrg 106>; + dmas = <&dma 20>, <&dma 21>; + dma-names = "rx","tx"; + #sound-dai-cells = <0>; + }; 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.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -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. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..d28c102c0ce7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,81 @@ +# 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: + - Andrew Davis <afd@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/tas2564 + https://www.ti.com/lit/gpn/tas2110 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,tas2562 + - 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': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt deleted file mode 100644 index ede6bb3d9637..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2770.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2770 Smart PA - -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. - -Required properties: - - - compatible: - Should contain "ti,tas2770". - - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. - -Optional properties: - -- interrupt-parent: the phandle to the interrupt controller which provides - the interrupt. -- interrupts: interrupt specification for data-ready. - -Examples: - - tas2770@4c { - compatible = "ti,tas2770"; - reg = <0x4c>; - #address-cells = <1>; - #size-cells = <0>; - interrupt-parent = <&msm_gpio>; - interrupts = <97 0>; - ti,asi-format = <0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..be2536e8c440 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,87 @@ +# 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. + +allOf: + - $ref: dai-common.yaml# + +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 + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@41 { + compatible = "ti,tas2770"; + reg = <0x41>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 14 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml new file mode 100644 index 000000000000..f2d878f6f495 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020-2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$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. + +allOf: + - $ref: dai-common.yaml# + +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': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <0>; + 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/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt index 7c8fd37c2f9e..1addc75989d5 100644 --- a/Documentation/devicetree/bindings/sound/tas571x.txt +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -12,6 +12,7 @@ Required properties: - "ti,tas5717", - "ti,tas5719", - "ti,tas5721" + - "ti,tas5733" - reg: The I2C address of the device - #sound-dai-cells: must be equal to 0 diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index 7481653fe8e3..7d851ae2bba2 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -4,13 +4,15 @@ 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/TAS5720A-Q1 +https://www.ti.com/product/TAS5722L Required properties: - compatible : "ti,tas5720", + "ti,tas5720a-q1", "ti,tas5722" - reg : I2C slave address - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry diff --git a/Documentation/devicetree/bindings/sound/tas5805m.yaml b/Documentation/devicetree/bindings/sound/tas5805m.yaml new file mode 100644 index 000000000000..12c41974274e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas5805m.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/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. + $ref: /schemas/types.yaml#/definitions/string + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + tas5805m: tas5805m@2c { + reg = <0x2c>; + compatible = "ti,tas5805m"; + + pvdd-supply = <&audiopwr>; + pdn-gpios = <&tlmm 160 0>; + + ti,dsp-config-name = "mono_pbtl_48khz"; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/test-component.yaml b/Documentation/devicetree/bindings/sound/test-component.yaml new file mode 100644 index 000000000000..9c40a2122dfd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/test-component.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/test-component.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Test Component + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - test-cpu + - test-cpu-verbose + - test-cpu-verbose-dai + - test-cpu-verbose-component + - test-codec + - test-codec-verbose + - test-codec-verbose-dai + - test-codec-verbose-component + +required: + - compatible + +additionalProperties: true + +examples: + - | + test_cpu { + compatible = "test-cpu"; + }; diff --git a/Documentation/devicetree/bindings/sound/tfa9879.txt b/Documentation/devicetree/bindings/sound/tfa9879.txt deleted file mode 100644 index 1620e6848436..000000000000 --- a/Documentation/devicetree/bindings/sound/tfa9879.txt +++ /dev/null @@ -1,23 +0,0 @@ -NXP TFA9879 class-D audio amplifier - -Required properties: - -- compatible : "nxp,tfa9879" - -- reg : the I2C address of the device - -- #sound-dai-cells : must be 0. - -Example: - -&i2c1 { - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2c1>; - - amp: amp@6c { - #sound-dai-cells = <0>; - compatible = "nxp,tfa9879"; - reg = <0x6c>; - }; -}; - diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml new file mode 100644 index 000000000000..20ea5883b7ff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml @@ -0,0 +1,139 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi <peter.ujfalusi@ti.com> +%YAML 1.2 +--- +$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..5b2874a80a4d --- /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, described 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 deleted file mode 100644 index a02ecaab5183..000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt +++ /dev/null @@ -1,56 +0,0 @@ -Texas Instruments pcm3168a DT bindings - -This driver supports both SPI and I2C bus access for this codec - -Required properties: - - - compatible: "ti,pcm3168a" - - - clocks : Contains an entry for each entry in clock-names - - - clock-names : Includes the following entries: - "scki" The system clock - - - VDD1-supply : Digital power supply regulator 1 (+3.3V) - - - VDD2-supply : Digital power supply regulator 2 (+3.3V) - - - VCCAD1-supply : ADC power supply regulator 1 (+5V) - - - VCCAD2-supply : ADC power supply regulator 2 (+5V) - - - VCCDA1-supply : DAC power supply regulator 1 (+5V) - - - VCCDA2-supply : DAC power supply regulator 2 (+5V) - -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 { - - ... - - 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>; - VDD2-supply = <&supply3v3>; - VCCAD1-supply = <&supply5v0>; - VCCAD2-supply = <&supply5v0>; - VCCDA1-supply = <&supply5v0>; - VCCDA2-supply = <&supply5v0>; - pinctrl-names = "default"; - pinctrl-0 = <&dac_clk_pin>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml b/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml new file mode 100644 index 000000000000..0b4f003989a4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm3168a.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM3168A Audio Codec + +maintainers: + - Damien Horsley <Damien.Horsley@imgtec.com> + - Geert Uytterhoeven <geert+renesas@glider.be> + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +description: + The Texas Instruments PCM3168A is a 24-bit Multi-channel Audio CODEC with + 96/192kHz sampling rate, supporting both SPI and I2C bus access. + +properties: + compatible: + const: ti,pcm3168a + + reg: + maxItems: 1 + + clocks: + items: + - description: System clock input + + clock-names: + items: + - const: scki + + reset-gpios: + items: + - description: | + GPIO line connected to the active-low RST pin of the codec. + RST = low: device power-down + RST = high: device is enabled + + "#sound-dai-cells": + enum: [0, 1] + + VDD1-supply: + description: Digital power supply regulator 1 (+3.3V) + + VDD2-supply: + description: Digital power supply regulator 2 (+3.3V) + + VCCAD1-supply: + description: ADC power supply regulator 1 (+5V) + + VCCAD2-supply: + description: ADC power supply regulator 2 (+5V) + + VCCDA1-supply: + description: DAC power supply regulator 1 (+5V) + + VCCDA2-supply: + description: DAC power supply regulator 2 (+5V) + + ports: + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + properties: + port@0: + $ref: audio-graph-port.yaml# + description: Audio input port. + + port@1: + $ref: audio-graph-port.yaml# + description: Audio output port. + +required: + - compatible + - reg + - clocks + - clock-names + - VDD1-supply + - VDD2-supply + - VCCAD1-supply + - VCCAD2-supply + - VCCDA1-supply + - VCCDA2-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm3168a: audio-codec@44 { + compatible = "ti,pcm3168a"; + reg = <0x44>; + reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>; + clocks = <&clk_core 42>; + clock-names = "scki"; + VDD1-supply = <&supply3v3>; + VDD2-supply = <&supply3v3>; + VCCAD1-supply = <&supply5v0>; + VCCAD2-supply = <&supply5v0>; + VCCDA1-supply = <&supply5v0>; + VCCDA2-supply = <&supply5v0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000000..27230c682d10 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax@flatmax.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..976238689249 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,122 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2563/TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + The TAS2563/TAS2781 is a mono, digital input Class-D audio + amplifier optimized for efficiently driving high peak power into + small loudspeakers. An integrated on-chip DSP supports Texas + Instruments Smart Amp speaker protection algorithm. The + 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/tas2563 + https://www.ti.com/lit/gpn/tas2781 + +properties: + compatible: + description: | + ti,tas2563: 6.1-W Boosted Class-D Audio Amplifier With Integrated + DSP and IV Sense, 16/20/24/32bit stereo I2S or multichannel TDM. + + ti,tas2781: 24-V Class-D Amplifier with Real Time Integrated Speaker + Protection and Audio Processing, 16/20/24/32bit stereo I2S or + multichannel TDM. + oneOf: + - items: + - enum: + - ti,tas2563 + - const: ti,tas2781 + - enum: + - ti,tas2781 + + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 8 + minItems: 1 + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,tas2563 + then: + properties: + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 4 + minItems: 1 + items: + minimum: 0x4c + maximum: 0x4f + + - if: + properties: + compatible: + contains: + enum: + - ti,tas2781 + then: + properties: + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 8 + minItems: 1 + items: + minimum: 0x38 + maximum: 0x3f + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad tas2781s, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad_tas2781: tas2781@38 { + compatible = "ti,tas2781"; + reg = <0x38>, /* Audio slot 0 */ + <0x3a>, /* Audio slot 1 */ + <0x39>, /* Audio slot 2 */ + <0x3b>; /* Audio slot 3 */ + + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + interrupt-parent = <&gpio1>; + interrupts = <15>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt index eacb54f34188..00940c489299 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas6424.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt @@ -19,4 +19,4 @@ tas6424: tas6424@6a { }; For more product information please see the link below: -http://www.ti.com/product/TAS6424-Q1 +https://www.ti.com/product/TAS6424-Q1 diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..ede14ca2c07a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,140 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 + +allOf: + - $ref: dai-common.yaml# + +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: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +unevaluatedProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: audio-codec@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml new file mode 100644 index 000000000000..4783e6dbb5c4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic32x4.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC32x4 Stereo Audio codec + +maintainers: + - Alexander Stein <alexander.stein@ew.tq-group.com> + +description: | + The TLV320AIC32x4 audio codec can be accessed using I2C or SPI + +properties: + compatible: + enum: + - ti,tas2505 + - ti,tlv320aic32x4 + - ti,tlv320aic32x6 + + reg: + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + av-supply: + description: Analog core power supply + + dv-supply: + description: Digital core power supply + + iov-supply: + description: Digital IO power supply + + ldoin-supply: + description: LDO power supply + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + aic32x4-gpio-func: + description: | + GPIO function configuration for pins MFP1-MFP5. + Types are defined in include/sound/tlv320aic32x4.h + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + - clocks + - clock-names + - iov-supply + +allOf: + - $ref: dai-common.yaml# + - if: + not: + required: + - ldoin-supply + then: + required: + - av-supply + - dv-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; + iov-supply = <®_3v3>; + ldoin-supply = <®_3v3>; + clocks = <&clks 201>; + clock-names = "mclk"; + aic32x4-gpio-func = < + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml new file mode 100644 index 000000000000..206f6d61e362 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -0,0 +1,166 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +# Copyright (C) 2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec + +description: | + TLV320AIC3x are a series of low-power stereo audio codecs with stereo + headphone amplifier, as well as multiple inputs and outputs programmable in + single-ended or fully differential configurations. + + The serial control bus supports SPI or I2C protocols, while the serial audio + data bus is programmable for I2S, left/right-justified, DSP, or TDM modes. + + The following pins can be referred in the sound node's audio routing property: + + CODEC output pins: + LLOUT + RLOUT + MONO_LOUT + HPLOUT + HPROUT + HPLCOM + HPRCOM + + CODEC input pins for TLV320AIC3104: + MIC2L + MIC2R + LINE1L + LINE1R + + CODEC input pins for other compatible codecs: + MIC3L + MIC3R + LINE1L + LINE2L + LINE1R + LINE2R + +maintainers: + - Jai Luthra <j-luthra@ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + gpio-reset: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + maxItems: 1 + description: + Deprecated, please use reset-gpios instead. + deprecated: true + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + + clocks: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: audio-codec@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: audio-codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml b/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml new file mode 100644 index 000000000000..785930658029 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,ts3a227e.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TS3A227E + Autonomous Audio Accessory Detection and Configuration Switch + +maintainers: + - Dylan Reid <dgreid@chromium.org> + +description: | + The TS3A227E detect headsets of 3-ring and 4-ring standards and + switches automatically to route the microphone correctly. It also + handles key press detection in accordance with the Android audio + headset specification v1.0. + +properties: + compatible: + enum: + - ti,ts3a227e + + reg: + const: 0x3b + + interrupts: + maxItems: 1 + + ti,micbias: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Intended MICBIAS voltage (datasheet section 9.6.7). + enum: + - 0 # 2.1 V + - 1 # 2.2 V + - 2 # 2.3 V + - 3 # 2.4 V + - 4 # 2.5 V + - 5 # 2.6 V + - 6 # 2.7 V + - 7 # 2.8 V + default: 1 + + ti,debounce-release-ms: + description: key release debounce time in ms (datasheet section 9.6.7). + enum: + - 0 + - 20 + default: 20 + + ti,debounce-press-ms: + description: key press debounce time in ms (datasheet section 9.6.7). + enum: + - 2 + - 40 + - 80 + - 120 + default: 80 + + ti,debounce-insertion-ms: + description: headset insertion debounce time in ms (datasheet section 9.6.5). + enum: + - 2 + - 30 + - 60 + - 90 + - 120 + - 150 + - 1000 + - 2000 + default: 90 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: audio-controller@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio1>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 2e6ac5d2ee96..f3274bcc4c05 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -8,7 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter maintainers: - - Dan Murphy <dmurphy@ti.com> + - Andrew Davis <afd@ti.com> description: | The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital @@ -18,46 +18,47 @@ description: | microphone bias or supply voltage generation. Specifications can be found at: - http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf - http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf - http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + 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: - oneOf: - - const: ti,tlv320adc3140 - - const: ti,tlv320adc5140 - - const: ti,tlv320adc6140 + 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 + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f reset-gpios: + maxItems: 1 description: | - GPIO used for hardware reset. + GPIO used for hardware reset. areg-supply: - description: | - Regulator with AVDD at 3.3V. If not defined then the internal regulator - is enabled. + 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 + 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 + 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] @@ -67,9 +68,9 @@ properties: 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. + channel is latched on the positive edge. 1 - Odd channel is latched on the positive edge and even channel is - latched on the the negative edge. + latched on 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 @@ -108,14 +109,90 @@ properties: maximum: 7 default: [0, 0, 0, 0] + 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 + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + + ti,asi-tx-drive: + type: boolean + description: | + When set the device will set the Tx ASI output to a Hi-Z state for unused + data cycles. Default is to drive the output low on unused ASI cycles. + +patternProperties: + '^ti,gpo-config-[1-4]$': + $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 + required: - compatible - reg +additionalProperties: false + examples: - | #include <dt-bindings/gpio/gpio.h> - i2c0 { + i2c { #address-cells = <1>; #size-cells = <0>; codec: codec@4c { @@ -124,6 +201,9 @@ examples: 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 e372303697dc..bbad98d5b986 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -58,7 +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 deleted file mode 100644 index ca75890f0d07..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ /dev/null @@ -1,41 +0,0 @@ -Texas Instruments - tlv320aic32x4 Codec module - -The tlv320aic32x4 serial control bus communicates through I2C protocols - -Required properties: - - compatible - "string" - One of: - "ti,tlv320aic32x4" TLV320AIC3204 - "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 - - reg: I2C slave address - - supply-*: Required supply regulators are: - "iov" - digital IO power supply - "ldoin" - LDO power supply - "dv" - Digital core power supply - "av" - Analog core power supply - If you supply ldoin, dv and av are optional. Otherwise they are required - See regulator/regulator.txt for more information about the detailed binding - format. - -Optional properties: - - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt - - clocks/clock-names: Clock named 'mclk' for the master clock of the codec. - See clock/clock-bindings.txt for information about the detailed format. - - aic32x4-gpio-func - <array of 5 int> - - Types are defined in include/sound/tlv320aic32x4.h - - -Example: - -codec: tlv320aic32x4@18 { - compatible = "ti,tlv320aic32x4"; - reg = <0x18>; - clocks = <&clks 201>; - clock-names = "mclk"; - aic32x4-gpio-func= < - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ - >; -}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 9796c4639262..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,80 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -Example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <®ulator>; - IOVDD-supply = <®ulator>; - DRVDD-supply = <®ulator>; - DVDD-supply = <®ulator>; -}; diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt deleted file mode 100644 index 21ab45bc7e8f..000000000000 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ /dev/null @@ -1,30 +0,0 @@ -Texas Instruments TS3A227E -Autonomous Audio Accessory Detection and Configuration Switch - -The TS3A227E detect headsets of 3-ring and 4-ring standards and -switches automatically to route the microphone correctly. It also -handles key press detection in accordance with the Android audio -headset specification v1.0. - -Required properties: - - - compatible: Should contain "ti,ts3a227e". - - reg: The i2c address. Should contain <0x3b>. - - interrupts: Interrupt number for /INT pin from the 227e - -Optional properies: - - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). - Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage - 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V - Default value is "1" (2.2V). - -Examples: - - i2c { - ts3a227e@3b { - compatible = "ti,ts3a227e"; - reg = <0x3b>; - interrupt-parent = <&gpio>; - interrupts = <3 IRQ_TYPE_LEVEL_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.yaml b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml index 22d54be7900a..8156f30eadd6 100644 --- a/Documentation/devicetree/bindings/sound/wlf,arizona.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml @@ -16,6 +16,9 @@ description: | This document lists sound specific bindings, see the primary binding document ../mfd/arizona.yaml +allOf: + - $ref: dai-common.yaml# + properties: '#sound-dai-cells': description: @@ -112,3 +115,5 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 1 maxItems: 12 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8510.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8510.yaml new file mode 100644 index 000000000000..6d12b0ac37e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8510.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8510.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8510 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8510 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8510"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8523.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8523.yaml new file mode 100644 index 000000000000..decc395bb873 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8523.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8523.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8523 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8523 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8523"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml new file mode 100644 index 000000000000..4d951ece394e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8524.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8524 24-bit 192KHz Stereo DAC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8524 + + "#sound-dai-cells": + const: 0 + + wlf,mute-gpios: + maxItems: 1 + description: + a GPIO spec for the MUTE pin. + +required: + - compatible + - wlf,mute-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + wm8524: codec { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8580.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8580.yaml new file mode 100644 index 000000000000..2f27852cdc20 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8580.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8580.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8580 and WM8581 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - wlf,wm8580 + - wlf,wm8581 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8580"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8711.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8711.yaml new file mode 100644 index 000000000000..ecaac2818b44 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8711.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8711.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8711 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8711 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8711"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8728.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8728.yaml new file mode 100644 index 000000000000..fc89475a051e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8728.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8728.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8728 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8728 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8728"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml new file mode 100644 index 000000000000..858c0f689581 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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. + +required: + - reg + - compatible + - AVDD-supply + - HPVDD-supply + - DBVDD-supply + - DCVDD-supply + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + wm8731_i2c: codec@0 { + compatible = "wlf,wm8731"; + reg = <0>; + spi-max-frequency = <12500000>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; + - | + + i2c { + #address-cells = <1>; + #size-cells = <0>; + wm8731_spi: codec@1b { + compatible = "wlf,wm8731"; + reg = <0x1b>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8737.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8737.yaml new file mode 100644 index 000000000000..12d8765726d8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8737.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8737.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8737 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8737 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8737"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8753.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8753.yaml new file mode 100644 index 000000000000..9eebe7d7f0b7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8753.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8753.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8753 audio CODEC + +description: | + Pins on the device (for linking into audio routes): + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * MONO1 + * MONO2 + * OUT3 + * OUT4 + * LINE1 + * LINE2 + * RXP + * RXN + * ACIN + * ACOP + * MIC1N + * MIC1 + * MIC2N + * MIC2 + * Mic Bias + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8753 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8753"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000000..4cfa66f62681 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml new file mode 100644 index 000000000000..329260cf0fa0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8904/WM8912 audio codecs + +maintainers: + - patches@opensource.cirrus.com + +description: | + Pins on the device (for linking into audio routes): + IN1L, IN1R, IN2L, IN2R, IN3L, IN3R, HPOUTL, HPOUTR, LINEOUTL, LINEOUTR, + MICBIAS + +properties: + compatible: + enum: + - wlf,wm8904 + - wlf,wm8912 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + AVDD-supply: true + CPVDD-supply: true + DBVDD-supply: true + DCVDD-supply: true + MICVDD-supply: true + +required: + - compatible + - reg + - clocks + - clock-names + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; + AVDD-supply = <®_1p8v>; + CPVDD-supply = <®_1p8v>; + DBVDD-supply = <®_1p8v>; + DCVDD-supply = <®_1p8v>; + MICVDD-supply = <®_1p8v>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml new file mode 100644 index 000000000000..3e809217c4ca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8940.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8940 Codec + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8940 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml new file mode 100644 index 000000000000..62e62c335d07 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8960 audio codec + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8960 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 0 + + AVDD-supply: + description: Analogue supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + SPKVDD1-supply: + description: Supply for speaker drivers 1. + + SPKVDD2-supply: + description: Supply for speaker drivers 2. + + wlf,capless: + type: boolean + description: + If present, OUT3 pin will be enabled and disabled together with HP_L and + HP_R pins in response to jack detect events. + + wlf,gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + description: | + A list of GPIO configuration register values. + - gpio-cfg[0]: ALRCGPIO of R9 (Audio interface) + - gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4). + + wlf,hp-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 3 + description: | + A list of headphone jack detect configuration register values: + - 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,shared-lrclk: + type: boolean + description: + If present, the LRCM bit of R24 (Additional control 2) gets set, + indicating that ADCLRC and DACLRC pins will be disabled only when ADC + (Left and Right) and DAC (Left and Right) are disabled. + When WM8960 works on synchronize mode and DACLRC pin is used to supply + frame clock, it will no frame clock for captrue unless enable DAC to + enable DACLRC pin. If shared-lrclk is present, no need to enable DAC for + captrue. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + clocks = <&clks 0>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + wlf,hp-cfg = <3 2 3>; + wlf,gpio-cfg = <1 3>; + wlf,shared-lrclk; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8961.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8961.yaml new file mode 100644 index 000000000000..f58078545569 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8961.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8961.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8961 Ultra-Low Power Stereo CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8961 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8961: codec@4a { + compatible = "wlf,wm8961"; + reg = <0x4a>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml new file mode 100644 index 000000000000..5fe0b2c9f99f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml @@ -0,0 +1,124 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 + +allOf: + - $ref: dai-common.yaml# + +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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + - PLLVDD-supply + - SPKVDD1-supply + - SPKVDD2-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/imx6qdl-clock.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8962: codec@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; + clocks = <&clks IMX6QDL_CLK_CKO>; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + CPVDD-supply = <®_audio>; + MICVDD-supply = <®_audio>; + PLLVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; + gpio-cfg = < + 0x0000 /* 0:Default */ + 0x0000 /* 1:Default */ + 0x0013 /* 2:FN_DMICCLK */ + 0x0000 /* 3:Default */ + 0x8014 /* 4:FN_DMICCDAT */ + 0x0000 /* 5:Default */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000000..efb5f9f6cc7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8994.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8994.yaml new file mode 100644 index 000000000000..8f045de02850 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8994.yaml @@ -0,0 +1,194 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8994.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM1811/WM8994/WM8958 audio codecs + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - patches@opensource.cirrus.com + +description: | + These devices support both I2C and SPI (configured with pin strapping on the + board). + + 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. + +properties: + compatible: + enum: + - wlf,wm1811 + - wlf,wm8994 + - wlf,wm8958 + + reg: + maxItems: 1 + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + minItems: 1 + items: + - const: MCLK1 + - const: MCLK2 + + gpio-controller: true + + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + interrupt-controller: true + + '#interrupt-cells': + const: 2 + description: + The first cell is the IRQ number. The second cell is the flags, encoded + as the trigger masks. + + AVDD1-supply: true + AVDD2-supply: true + CPVDD-supply: true + DBVDD-supply: true + DBVDD1-supply: true + DBVDD2-supply: true + DBVDD3-supply: true + DCVDD-supply: true + LDO1VDD-supply: true + LDO2VDD-supply: true + SPKVDD1-supply: true + SPKVDD2-supply: true + + '#sound-dai-cells': + const: 0 + + wlf,gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 11 + description: + A list of GPIO configuration register values. If absent, no configuration + of these registers is performed. If any value is over 0xffff then the + register will be left as default. If present 11 values must be supplied. + + wlf,micbias-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + description: + Two MICBIAS register values for WM1811 or WM8958. If absent the register + defaults will be used. + + wlf,ldo1ena-gpios: + maxItems: 1 + description: + Control of LDO1ENA input to device. + + wlf,ldo2ena-gpios: + maxItems: 1 + description: + Control of LDO2ENA input to device. + + wlf,lineout1-se: + type: boolean + description: + LINEOUT1 is in single ended mode. + + wlf,lineout2-se: + type: boolean + description: + INEOUT2 is in single ended mode. + + wlf,lineout1-feedback: + type: boolean + description: + LINEOUT1 has common mode feedback connected. + + wlf,lineout2-feedback: + type: boolean + description: + LINEOUT2 has common mode feedback connected. + + wlf,ldoena-always-driven: + type: boolean + description: + LDOENA is always driven. + + wlf,spkmode-pu: + type: boolean + description: + Enable the internal pull-up resistor on the SPKMODE pin. + + wlf,csnaddr-pd: + type: boolean + description: + Enable the internal pull-down resistor on the CS/ADDR pin. + +required: + - compatible + - reg + - AVDD2-supply + - CPVDD-supply + - SPKVDD1-supply + - SPKVDD2-supply + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + enum: + - wlf,wm1811 + - wlf,wm8958 + then: + properties: + DBVDD-supply: false + LDO2VDD-supply: false + required: + - DBVDD1-supply + - DBVDD2-supply + - DBVDD3-supply + else: + properties: + DBVDD1-supply: false + DBVDD2-supply: false + DBVDD3-supply: false + required: + - DBVDD-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1a { + compatible = "wlf,wm1811"; + reg = <0x1a>; + clocks = <&i2s0 0>; + clock-names = "MCLK1"; + + AVDD2-supply = <&main_dc_reg>; + CPVDD-supply = <&main_dc_reg>; + DBVDD1-supply = <&main_dc_reg>; + DBVDD2-supply = <&main_dc_reg>; + DBVDD3-supply = <&main_dc_reg>; + LDO1VDD-supply = <&main_dc_reg>; + SPKVDD1-supply = <&main_dc_reg>; + SPKVDD2-supply = <&main_dc_reg>; + + wlf,ldo1ena-gpios = <&gpb0 0 GPIO_ACTIVE_HIGH>; + wlf,ldo2ena-gpios = <&gpb0 1 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8510.txt b/Documentation/devicetree/bindings/sound/wm8510.txt deleted file mode 100644 index e6b6cc041f89..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8510.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8510 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8510" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8510: codec@1a { - compatible = "wlf,wm8510"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8523.txt b/Documentation/devicetree/bindings/sound/wm8523.txt deleted file mode 100644 index f3a6485f4b8a..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8523.txt +++ /dev/null @@ -1,16 +0,0 @@ -WM8523 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8523" - - - reg : the I2C address of the device. - -Example: - -wm8523: codec@1a { - compatible = "wlf,wm8523"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt deleted file mode 100644 index f6c0c263b135..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8524.txt +++ /dev/null @@ -1,16 +0,0 @@ -WM8524 audio CODEC - -This device does not use I2C or SPI but a simple Hardware Control Interface. - -Required properties: - - - compatible : "wlf,wm8524" - - - wlf,mute-gpios: a GPIO spec for the MUTE pin. - -Example: - -wm8524: codec { - compatible = "wlf,wm8524"; - wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8580.txt b/Documentation/devicetree/bindings/sound/wm8580.txt deleted file mode 100644 index ff3f9f5f2111..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8580.txt +++ /dev/null @@ -1,16 +0,0 @@ -WM8580 and WM8581 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8580", "wlf,wm8581" - - - reg : the I2C address of the device. - -Example: - -wm8580: codec@1a { - compatible = "wlf,wm8580"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8711.txt b/Documentation/devicetree/bindings/sound/wm8711.txt deleted file mode 100644 index c30a1387c4bf..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8711.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8711 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8711" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8711: codec@1a { - compatible = "wlf,wm8711"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8728.txt b/Documentation/devicetree/bindings/sound/wm8728.txt deleted file mode 100644 index a3608b4c78b9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8728.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8728 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8728" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8728: codec@1a { - compatible = "wlf,wm8728"; - 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/wm8737.txt b/Documentation/devicetree/bindings/sound/wm8737.txt deleted file mode 100644 index eda1ec6a7563..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8737.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8737 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8737" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8737: codec@1a { - compatible = "wlf,wm8737"; - reg = <0x1a>; -}; 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/wm8753.txt b/Documentation/devicetree/bindings/sound/wm8753.txt deleted file mode 100644 index eca9e5a825a9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8753.txt +++ /dev/null @@ -1,40 +0,0 @@ -WM8753 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8753" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Pins on the device (for linking into audio routes): - - * LOUT1 - * LOUT2 - * ROUT1 - * ROUT2 - * MONO1 - * MONO2 - * OUT3 - * OUT4 - * LINE1 - * LINE2 - * RXP - * RXN - * ACIN - * ACOP - * MIC1N - * MIC1 - * MIC2N - * MIC2 - * Mic Bias - -Example: - -wm8753: codec@1a { - compatible = "wlf,wm8753"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8782.txt b/Documentation/devicetree/bindings/sound/wm8782.txt index 256cdec6ec4d..1a28f3280972 100644 --- a/Documentation/devicetree/bindings/sound/wm8782.txt +++ b/Documentation/devicetree/bindings/sound/wm8782.txt @@ -8,10 +8,17 @@ Required properties: - Vdda-supply : phandle to a regulator for the analog power supply (2.7V - 5.5V) - Vdd-supply : phandle to a regulator for the digital power supply (2.7V - 3.6V) +Optional properties: + + - wlf,fsampen: + FSAMPEN pin value, 0 for low, 1 for high, 2 for disconnected. + Defaults to 0 if left unspecified. + Example: wm8782: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&vdda_supply>; Vdd-supply = <&vdd_supply>; + wlf,fsampen = <2>; /* 192KHz */ }; 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/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt deleted file mode 100644 index 66bf261423b9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8904.txt +++ /dev/null @@ -1,33 +0,0 @@ -WM8904 audio CODEC - -This device supports I2C only. - -Required properties: - - compatible: "wlf,wm8904" or "wlf,wm8912" - - reg: the I2C address of the device. - - clock-names: "mclk" - - clocks: reference to - <Documentation/devicetree/bindings/clock/clock-bindings.txt> - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - -Examples: - -codec: wm8904@1a { - compatible = "wlf,wm8904"; - reg = <0x1a>; - clocks = <&pck0>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt deleted file mode 100644 index 6d29ac3750ee..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ /dev/null @@ -1,31 +0,0 @@ -WM8960 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8960" - - - reg : the I2C address of the device. - -Optional properties: - - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of - R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins - will be disabled only when ADC (Left and Right) and DAC (Left and Right) - are disabled. - When wm8960 works on synchronize mode and DACLRC pin is used to supply - frame clock, it will no frame clock for captrue unless enable DAC to enable - DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue. - - - wlf,capless: This is a boolean property. If present, OUT3 pin will be - enabled and disabled together with HP_L and HP_R pins in response to jack - detect events. - -Example: - -wm8960: codec@1a { - compatible = "wlf,wm8960"; - reg = <0x1a>; - - wlf,shared-lrclk; -}; 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 deleted file mode 100644 index 367b58ce1bb9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ /dev/null @@ -1,89 +0,0 @@ -WM1811/WM8994/WM8958 audio CODEC - -These devices support both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : One of "wlf,wm1811", "wlf,wm8994" or "wlf,wm8958". - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - - - gpio-controller : Indicates this device is a GPIO controller. - - #gpio-cells : Must be 2. The first cell is the pin number and the - second cell is used to specify optional parameters (currently unused). - - - 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: - - - interrupts : The interrupt line the IRQ signal for the device is - connected to. This is optional, if it is not connected then none - of the interrupt related properties should be specified. - - interrupt-controller : These devices contain interrupt controllers - and may provide interrupt services to other devices if they have an - interrupt line connected. - - #interrupt-cells: the number of cells to describe an IRQ, this should be 2. - The first cell is the IRQ number. - The second cell is the flags, encoded as the trigger masks from - Documentation/devicetree/bindings/interrupt-controller/interrupts.txt - - - clocks : A list of up to two phandle and clock specifier pairs - - clock-names : A list of clock names sorted in the same order as clocks. - Valid clock names are "MCLK1" and "MCLK2". - - - wlf,gpio-cfg : A list of GPIO configuration register values. If absent, - no configuration of these registers is performed. If any value is - over 0xffff then the register will be left as default. If present 11 - values must be supplied. - - - wlf,micbias-cfg : Two MICBIAS register values for WM1811 or - WM8958. If absent the register defaults will be used. - - - wlf,ldo1ena : GPIO specifier for control of LDO1ENA input to device. - - wlf,ldo2ena : GPIO specifier for control of LDO2ENA input to device. - - - wlf,lineout1-se : If present LINEOUT1 is in single ended mode. - - wlf,lineout2-se : If present LINEOUT2 is in single ended mode. - - - wlf,lineout1-feedback : If present LINEOUT1 has common mode feedback - connected. - - wlf,lineout2-feedback : If present LINEOUT2 has common mode feedback - connected. - - - wlf,ldoena-always-driven : If present LDOENA is always driven. - - - wlf,spkmode-pu : If present enable the internal pull-up resistor on - the SPKMODE pin. - - - wlf,csnaddr-pd : If present enable the internal pull-down resistor on - the CS/ADDR pin. - -Example: - -wm8994: codec@1a { - compatible = "wlf,wm8994"; - reg = <0x1a>; - - gpio-controller; - #gpio-cells = <2>; - - lineout1-se; - - AVDD1-supply = <®ulator>; - AVDD2-supply = <®ulator>; - CPVDD-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 index 338e2a13c775..8bd201e573aa 100644 --- a/Documentation/devicetree/bindings/sound/zl38060.yaml +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -15,6 +15,9 @@ maintainers: - Jaroslav Kysela <perex@perex.cz> - Takashi Iwai <tiwai@suse.com> +allOf: + - $ref: dai-common.yaml# + properties: compatible: const: mscc,zl38060 @@ -48,12 +51,12 @@ required: - gpio-controller - '#sound-dai-cells' -additionalProperties: false +unevaluatedProperties: false examples: - | #include <dt-bindings/gpio/gpio.h> - spi0 { + spi { #address-cells = <1>; #size-cells = <0>; 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"; - }; |