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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 18:49:45 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 18:49:45 +0000 |
| commit | 2c3c1048746a4622d8c89a29670120dc8fab93c4 (patch) | |
| tree | 848558de17fb3008cdf4d861b01ac7781903ce39 /Documentation/devicetree/bindings/sound | |
| parent | Initial commit. (diff) | |
| download | linux-2c3c1048746a4622d8c89a29670120dc8fab93c4.tar.xz linux-2c3c1048746a4622d8c89a29670120dc8fab93c4.zip | |
Adding upstream version 6.1.76.upstream/6.1.76
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'Documentation/devicetree/bindings/sound')
386 files changed, 26315 insertions, 0 deletions
diff --git a/Documentation/devicetree/bindings/sound/ac97-bus.txt b/Documentation/devicetree/bindings/sound/ac97-bus.txt new file mode 100644 index 000000000..103c428f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ac97-bus.txt @@ -0,0 +1,32 @@ +Generic AC97 Device Properties + +This documents describes the devicetree bindings for an ac97 controller child +node describing ac97 codecs. + +Required properties: +-compatible : Must be "ac97,vendor_id1,vendor_id2 + The ids shall be the 4 characters hexadecimal encoding, such as + given by "%04x" formatting of printf +-reg : Must be the ac97 codec number, between 0 and 3 + +Example: +ac97: sound@40500000 { + compatible = "marvell,pxa270-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pinctrl_ac97_default >; + clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>; + clock-names = "AC97CLK", "AC97CONFCLK"; + + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + reg = <0>; + compatible = "ac97,574d,4c13"; + clocks = <&fixed_wm9713_clock>; + clock-names = "ac97_clk"; + } +}; diff --git a/Documentation/devicetree/bindings/sound/adi,adau1372.yaml b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml new file mode 100644 index 000000000..59f7c60a1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1372.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + + +title: Analog Devices ADAU1372 CODEC + +maintainers: + - Alexandre Belloni <alexandre.belloni@bootlin.om> + +description: | + Analog Devices ADAU1372 four inputs and two outputs codec. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf + +properties: + compatible: + enum: + - adi,adau1372 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + powerdown-gpios: + description: GPIO used for hardware power-down. + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@3c { + compatible = "adi,adau1372"; + reg = <0x3c>; + #sound-dai-cells = <0>; + clock-names = "mclk"; + clocks = <&adau1372z_xtal>; + }; + }; + + adau1372z_xtal: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <12288000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt new file mode 100644 index 000000000..0d1128ce2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -0,0 +1,39 @@ +Analog Devices ADAU1701 + +Required properties: + + - compatible: Should contain "adi,adau1701" + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + - adi,pll-mode-gpios: An array of two GPIO specs to describe the GPIOs + the ADAU's PLL config pins are connected to. + The state of the pins are set according to the + configured clock divider on ASoC side before the + firmware is loaded. + - adi,pin-config: An array of 12 numerical values selecting one of the + pin configurations as described in the datasheet, + table 53. Note that the value of this property has + to be prefixed with '/bits/ 8'. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + +Examples: + + i2c_bus { + adau1701@34 { + compatible = "adi,adau1701"; + reg = <0x34>; + reset-gpio = <&gpio 23 0>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; + adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>; + adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4 + 0x4 0x4 0x4 0x4 0x4 0x4>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt new file mode 100644 index 000000000..1447dec28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -0,0 +1,32 @@ +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,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml new file mode 100644 index 000000000..847b83398 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1977.yaml# +$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: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + spi { + #address-cells = <1>; + #size-cells = <0>; + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + reg = <0>; + spi-max-frequency = <600000>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + + adi,micbias = <3>; + }; + }; + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt new file mode 100644 index 000000000..f144ee1ab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.txt @@ -0,0 +1,19 @@ +Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +Required properties: + + - compatible: Must be "adi,adau7002" + +Optional properties: + + - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD + supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt + + If this property is not present it is assumed that the supply pin is + hardwired to always on. + +Example: + adau7002: pdm-to-i2s { + compatible = "adi,adau7002"; + IOVDD-supply = <&supply>; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml new file mode 100644 index 000000000..fb78967ee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7118.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + + +title: Analog Devices ADAU7118 8 Channel PDM to I2S/TDM Converter + +maintainers: + - Nuno Sá <nuno.sa@analog.com> + +description: | + Analog Devices ADAU7118 8 Channel PDM to I2S/TDM Converter over I2C or HW + standalone mode. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU7118.pdf + +properties: + compatible: + enum: + - adi,adau7118 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + iovdd-supply: + description: Digital Input/Output Power Supply. + + dvdd-supply: + description: Internal Core Digital Power Supply. + + adi,decimation-ratio: + description: | + This property set's the decimation ratio of PDM to PCM audio data. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [64, 32, 16] + default: 64 + + adi,pdm-clk-map: + description: | + The ADAU7118 has two PDM clocks for the four Inputs. Each input must be + assigned to one of these two clocks. This property set's the mapping + between the clocks and the inputs. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + items: + maximum: 1 + default: [0, 0, 1, 1] + +required: + - "#sound-dai-cells" + - compatible + - iovdd-supply + - dvdd-supply + +additionalProperties: false + +examples: + - | + i2c { + /* example with i2c support */ + #address-cells = <1>; + #size-cells = <0>; + adau7118_codec: audio-codec@14 { + compatible = "adi,adau7118"; + reg = <0x14>; + #sound-dai-cells = <0>; + iovdd-supply = <&supply>; + dvdd-supply = <&supply>; + adi,pdm-clk-map = <1 1 0 0>; + adi,decimation-ratio = <16>; + }; + }; + + /* example with hw standalone mode */ + adau7118_codec_hw: adau7118-codec-hw { + compatible = "adi,adau7118"; + #sound-dai-cells = <0>; + iovdd-supply = <&supply>; + dvdd-supply = <&supply>; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 000000000..229ad1392 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,34 @@ +ADI AXI-I2S controller + +The core can be generated with transmit (playback), only receive +(capture) or both directions enabled. + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" 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 two dma channels if both transmit and receive are + enabled, one channel otherwise. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +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: + + i2s: i2s@77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>, <&ps7_dma 1>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt new file mode 100644 index 000000000..7b664e7cb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt @@ -0,0 +1,30 @@ +ADI AXI-SPDIF controller + +Required properties: + - compatible : Must be "adi,axi-spdif-tx-1.00.a" + - reg : Must contain SPDIF core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names: "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - 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: + + spdif: spdif@77400000 { + compatible = "adi,axi-spdif-tx-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>; + dma-names = "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml new file mode 100644 index 000000000..fd5aa61b4 --- /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,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt new file mode 100644 index 000000000..a9c9d83c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,14 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + - shutdown-gpios : The gpio connected to the shutdown pin. + The gpio signal is ACTIVE_LOW. + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt new file mode 100644 index 000000000..3b3302fe3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt @@ -0,0 +1,19 @@ +Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices + +SSM2602 support both I2C and SPI as the configuration interface, +the selection is made by the MODE strap-in pin. +SSM2603 and SSM2604 only support I2C as the configuration interface. + +Required properties: + + - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + Example: + + ssm2602: ssm2602@1a { + compatible = "adi,ssm2602"; + reg = <0x1a>; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4104.txt b/Documentation/devicetree/bindings/sound/ak4104.txt new file mode 100644 index 000000000..ae5f7f057 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4104.txt @@ -0,0 +1,25 @@ +AK4104 S/PDIF transmitter + +This device supports SPI mode only. + +Required properties: + + - compatible : "asahi-kasei,ak4104" + + - reg : The chip select number on the SPI bus + + - vdd-supply : A regulator node, providing 2.7V - 3.6V + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the device starts. + +Example: + +spdif: ak4104@0 { + compatible = "asahi-kasei,ak4104"; + reg = <0>; + spi-max-frequency = <5000000>; + vdd-supply = <&vdd_3v3_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/ak4118.txt b/Documentation/devicetree/bindings/sound/ak4118.txt new file mode 100644 index 000000000..6e11a2f74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4118.txt @@ -0,0 +1,22 @@ +AK4118 S/PDIF transceiver + +This device supports I2C mode. + +Required properties: + +- compatible : "asahi-kasei,ak4118" +- reg : The I2C address of the device for I2C +- reset-gpios: A GPIO specifier for the reset pin +- irq-gpios: A GPIO specifier for the IRQ pin + +Example: + +&i2c { + ak4118: ak4118@13 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4118"; + reg = <0x13>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW> + irq-gpios = <&gpio 1 GPIO_ACTIVE_HIGH>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/ak4375.yaml b/Documentation/devicetree/bindings/sound/ak4375.yaml new file mode 100644 index 000000000..5f0fc584b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4375.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4375 DAC and headphones amplifier + +maintainers: + - Vincent Knecht <vincent.knecht@mailoo.org> + +properties: + compatible: + const: asahi-kasei,ak4375 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + avdd-supply: + description: regulator phandle for the AVDD power supply. + + tvdd-supply: + description: regulator phandle for the TVDD power supply. + + pdn-gpios: + description: optional GPIO to set the PDN pin. + +required: + - compatible + - reg + - '#sound-dai-cells' + - avdd-supply + - tvdd-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + headphones: audio-codec@10 { + compatible = "asahi-kasei,ak4375"; + reg = <0x10>; + avdd-supply = <®_headphones_avdd>; + tvdd-supply = <&pm8916_l6>; + pdn-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphones_pdn_default>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt new file mode 100644 index 000000000..0416c1489 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -0,0 +1,28 @@ +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 +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply +- dsd-path: Select DSD input pins for ak4497 + 0: select #16, #17, #19 pins + 1: select #3, #4, #5 pins + +Example: + +&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/ak4554.txt b/Documentation/devicetree/bindings/sound/ak4554.txt new file mode 100644 index 000000000..934fa0275 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4554.txt @@ -0,0 +1,11 @@ +AK4554 ADC/DAC + +Required properties: + + - compatible : "asahi-kasei,ak4554" + +Example: + +ak4554-adc-dac { + compatible = "asahi-kasei,ak4554"; +}; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml new file mode 100644 index 000000000..aa8a258a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: codec@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml new file mode 100644 index 000000000..48a5b2c39 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4642 + - asahi-kasei,ak4643 + - asahi-kasei,ak4648 + + reg: + maxItems: 1 + + "#clock-cells": + const: 0 + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-frequency: + description: common clock binding; frequency of MCKO + + clock-output-names: + description: common clock name + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + #sound-dai-cells = <0>; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt new file mode 100644 index 000000000..ec3df3abb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -0,0 +1,23 @@ +AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + +This device has no control interface. + +Required properties: + + - compatible : "asahi-kasei,ak5386" + +Optional properties: + + - reset-gpio : a GPIO spec for the reset/power down pin. + If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V + +Example: + +spdif: ak5386@0 { + compatible = "asahi-kasei,ak5386"; + reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/ak5558.txt b/Documentation/devicetree/bindings/sound/ak5558.txt new file mode 100644 index 000000000..e28708db6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5558.txt @@ -0,0 +1,24 @@ +AK5558 8 channel differential 32-bit delta-sigma ADC + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak5558" or "asahi-kasei,ak5552". +- reg : The I2C address of the device. + +Optional properties: + +- reset-gpios: A GPIO specifier for the power down & reset pin. +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply + +Example: + +&i2c { + ak5558: adc@10 { + compatible = "asahi-kasei,ak5558"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/alc5623.txt b/Documentation/devicetree/bindings/sound/alc5623.txt new file mode 100644 index 000000000..26c86c98d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/alc5623.txt @@ -0,0 +1,25 @@ +ALC5621/ALC5622/ALC5623 audio Codec + +Required properties: + + - compatible: "realtek,alc5623" + - reg: the I2C address of the device. + +Optional properties: + + - add-ctrl: Default register value for Reg-40h, Additional Control + Register. If absent or has the value of 0, the + register is untouched. + + - jack-det-ctrl: Default register value for Reg-5Ah, Jack Detect + Control Register. If absent or has value 0, the + register is untouched. + +Example: + + alc5621: alc5621@1a { + compatible = "alc5621"; + reg = <0x1a>; + add-ctrl = <0x3700>; + jack-det-ctrl = <0x4810>; + }; diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt new file mode 100644 index 000000000..ffd886d11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -0,0 +1,43 @@ +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 new file mode 100644 index 000000000..292fcb643 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,267 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + maxItems: 18 + items: + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + maxItems: 1 + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml new file mode 100644 index 000000000..dd30881ad --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml @@ -0,0 +1,146 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 I2S Controller + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: allwinner,sun4i-a10-i2s + - 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 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: mod + + # Even though it only applies to subschemas under the conditionals, + # not listing them here will trigger a warning because of the + # additionalsProperties set to false. + dmas: true + dma-names: true + resets: + maxItems: 1 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-i2s + - allwinner,sun8i-a83t-i2s + - allwinner,sun8i-h3-i2s + - allwinner,sun50i-a64-codec-i2s + - allwinner,sun50i-h6-i2s + - allwinner,sun50i-r329-i2s + + then: + required: + - resets + + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun8i-a83t-i2s + - allwinner,sun8i-h3-i2s + + then: + properties: + dmas: + minItems: 1 + items: + - description: RX DMA Channel + - description: TX DMA Channel + description: + Some controllers cannot receive but can only transmit + data. In such a case, the RX DMA channel is to be omitted. + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - const: tx + description: + Some controllers cannot receive but can only transmit + data. In such a case, the RX name is to be omitted. + + else: + properties: + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + i2s0: i2s@1c22400 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-i2s"; + reg = <0x01c22400 0x400>; + interrupts = <0 16 4>; + clocks = <&apb0_gates 3>, <&i2s0_clk>; + clock-names = "apb", "mod"; + dmas = <&dma 0 3>, <&dma 0 3>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml new file mode 100644 index 000000000..68c84e29c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -0,0 +1,122 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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 + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Liam Girdwood <lgirdwood@gmail.com> + - Mark Brown <broonie@kernel.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: allwinner,sun4i-a10-spdif + - const: allwinner,sun6i-a31-spdif + - const: allwinner,sun8i-h3-spdif + - const: allwinner,sun50i-h6-spdif + - items: + - const: allwinner,sun8i-a83t-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun50i-a64-spdif + - const: allwinner,sun8i-h3-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: spdif + + # Even though it only applies to subschemas under the conditionals, + # not listing them here will trigger a warning because of the + # additionalsProperties set to false. + dmas: true + dma-names: true + resets: + maxItems: 1 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-spdif + - allwinner,sun8i-h3-spdif + + then: + required: + - resets + + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif + + then: + properties: + dmas: + description: TX DMA Channel + + dma-names: + const: tx + + else: + properties: + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + spdif: spdif@1c21000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-spdif"; + reg = <0x01c21000 0x40>; + interrupts = <13>; + clocks = <&apb0_gates 1>, <&spdif_clk>; + clock-names = "apb", "spdif"; + dmas = <&dma 0 2>, <&dma 0 2>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml new file mode 100644 index 000000000..5800de63f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-a64-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A64 Analog Codec + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + compatible: + const: allwinner,sun50i-a64-codec-analog + + reg: + maxItems: 1 + + cpvdd-supply: + description: + Regulator for the headphone amplifier + + allwinner,internal-bias-resistor: + description: + Enable the internal 2.2K bias resistor between HBIAS and MICDET pins + type: boolean + +required: + - compatible + - reg + - cpvdd-supply + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun50i-a64-codec-analog"; + reg = <0x01f015c0 0x4>; + cpvdd-supply = <®_eldo1>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000..2f12cabe4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC + +maintainers: + - Ban Tao <fengzheng923@gmail.com> + +properties: + compatible: + const: allwinner,sun50i-h6-dmic + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml new file mode 100644 index 000000000..1c21a1b39 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a23-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A23 Analog Codec + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + compatible: + oneOf: + # FIXME: This is documented in the PRCM binding, but needs to be + # migrated here at some point + # - allwinner,sun8i-a23-codec-analog + - const: allwinner,sun8i-h3-codec-analog + - items: + - const: allwinner,sun8i-v3-codec-analog + - const: allwinner,sun8i-h3-codec-analog + - const: allwinner,sun8i-v3s-codec-analog + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun8i-h3-codec-analog"; + reg = <0x01f015c0 0x4>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml new file mode 100644 index 000000000..4eb11a8e6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a33-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A33 Codec + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + "#sound-dai-cells": + minimum: 0 + maximum: 1 + description: + A value of 0 is deprecated. When used, it only allows access to + the ADC/DAC and AIF1 (the CPU DAI), not the other two AIFs/DAIs. + + compatible: + oneOf: + - items: + - const: allwinner,sun50i-a64-codec + - const: allwinner,sun8i-a33-codec + - const: allwinner,sun8i-a33-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + audio-codec@1c22e00 { + #sound-dai-cells = <1>; + compatible = "allwinner,sun8i-a33-codec"; + reg = <0x01c22e00 0x400>; + interrupts = <0 29 4>; + clocks = <&ccu 47>, <&ccu 92>; + clock-names = "bus", "mod"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml new file mode 100644 index 000000000..0705f9119 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -0,0 +1,118 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,aiu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AIU audio output controller + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 2 + + compatible: + items: + - enum: + - amlogic,aiu-gxbb + - amlogic,aiu-gxl + - amlogic,aiu-meson8 + - amlogic,aiu-meson8b + - const: amlogic,aiu + + clocks: + items: + - description: AIU peripheral clock + - description: I2S peripheral clock + - description: I2S output clock + - description: I2S master clock + - description: I2S mixer clock + - description: SPDIF peripheral clock + - description: SPDIF output clock + - description: SPDIF master clock + - description: SPDIF master clock multiplexer + + clock-names: + items: + - const: pclk + - const: i2s_pclk + - const: i2s_aoclk + - const: i2s_mclk + - const: i2s_mixer + - const: spdif_pclk + - const: spdif_aoclk + - const: spdif_mclk + - const: spdif_mclk_sel + + interrupts: + items: + - description: I2S interrupt line + - description: SPDIF interrupt line + + interrupt-names: + items: + - const: i2s + - const: spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - clocks + - clock-names + - interrupts + - interrupt-names + - reg + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/gxbb-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/reset/amlogic,meson-gxbb-reset.h> + + aiu: audio-controller@5400 { + compatible = "amlogic,aiu-gxl", "amlogic,aiu"; + #sound-dai-cells = <2>; + reg = <0x5400 0x2ac>; + interrupts = <GIC_SPI 48 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 50 IRQ_TYPE_EDGE_RISING>; + interrupt-names = "i2s", "spdif"; + clocks = <&clkc CLKID_AIU_GLUE>, + <&clkc CLKID_I2S_OUT>, + <&clkc CLKID_AOCLK_GATE>, + <&clkc CLKID_CTS_AMCLK>, + <&clkc CLKID_MIXER_IFACE>, + <&clkc CLKID_IEC958>, + <&clkc CLKID_IEC958_GATE>, + <&clkc CLKID_CTS_MCLK_I958>, + <&clkc CLKID_CTS_I958>; + clock-names = "pclk", + "i2s_pclk", + "i2s_aoclk", + "i2s_mclk", + "i2s_mixer", + "spdif_pclk", + "spdif_aoclk", + "spdif_mclk", + "spdif_mclk_sel"; + resets = <&reset RESET_AIU>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt new file mode 100644 index 000000000..fa4545ed8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt @@ -0,0 +1,34 @@ +* 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-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt new file mode 100644 index 000000000..716878107 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt @@ -0,0 +1,29 @@ +* 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-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt new file mode 100644 index 000000000..80b411296 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt @@ -0,0 +1,124 @@ +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-spdifin.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt new file mode 100644 index 000000000..df92a4ecf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt @@ -0,0 +1,27 @@ +* 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-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt new file mode 100644 index 000000000..28381dd1f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt @@ -0,0 +1,25 @@ +* 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-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt new file mode 100644 index 000000000..5996c0cd8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt @@ -0,0 +1,36 @@ +* 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-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt new file mode 100644 index 000000000..cabfb26a5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt @@ -0,0 +1,22 @@ +* 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,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml new file mode 100644 index 000000000..77469a45b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a Internal DAC Control Glue + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: amlogic,g12a-toacodec + - items: + - enum: + - amlogic,sm1-toacodec + - const: amlogic,g12a-toacodec + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + toacodec: audio-controller@740 { + compatible = "amlogic,g12a-toacodec"; + reg = <0x740 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOACODEC>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt new file mode 100644 index 000000000..4e8cd7eb7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt @@ -0,0 +1,58 @@ +* Amlogic HDMI Tx control glue + +Required properties: +- compatible: "amlogic,g12a-tohdmitx" or + "amlogic,sm1-tohdmitx" +- reg: physical base address of the controller and length of memory + mapped region. +- #sound-dai-cells: should be 1. +- resets: phandle to the dedicated reset line of the hdmitx glue. + +Example on the S905X2 SoC: + +tohdmitx: audio-controller@744 { + compatible = "amlogic,g12a-tohdmitx"; + reg = <0x0 0x744 0x0 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOHDMITX>; +}; + +Example of an 'amlogic,axg-sound-card': + +sound { + compatible = "amlogic,axg-sound-card"; + +[...] + + dai-link-x { + sound-dai = <&tdmif_a>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-0 = <1 1>; + + codec-0 { + sound-dai = <&tohdmitx TOHDMITX_I2S_IN_A>; + }; + + codec-1 { + sound-dai = <&external_dac>; + }; + }; + + dai-link-y { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-0 = <1 1>; + + codec { + sound-dai = <&tohdmitx TOHDMITX_I2S_IN_C>; + }; + }; + + dai-link-z { + sound-dai = <&tohdmitx TOHDMITX_I2S_OUT>; + + codec { + sound-dai = <&hdmi_tx>; + }; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml new file mode 100644 index 000000000..b358fd601 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic GX sound card + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + items: + - const: amlogic,gx-sound-card + + audio-aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: list of auxiliary devices + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + description: |- + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + description: |- + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +patternProperties: + "^dai-link-[0-9]+$": + type: object + additionalProperties: false + description: |- + dai-link child nodes: + Container for dai-link level properties and the CODEC sub-nodes. + There should be at least one (and probably more) subnode of this type + + properties: + dai-format: + $ref: /schemas/types.yaml#/definitions/string + enum: [ i2s, left-j, dsp_a ] + + mclk-fs: + $ref: /schemas/types.yaml#/definitions/uint32 + description: |- + Multiplication factor between the frame rate and master clock + rate + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + patternProperties: + "^codec(-[0-9]+)?$": + type: object + additionalProperties: false + description: |- + Codecs: + dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. dai-link representing + frontend links have no codec, therefore have no subnodes + + properties: + sound-dai: + maxItems: 1 + description: phandle of the codec DAI + + required: + - sound-dai + + required: + - sound-dai + +required: + - model + - dai-link-0 + +additionalProperties: false + +examples: + - | + sound { + compatible = "amlogic,gx-sound-card"; + model = "GXL-ACME-S905X-FOO"; + audio-aux-devs = <&>; + audio-routing = "I2S ENCODER I2S IN", "I2S FIFO Playback"; + + dai-link-0 { + sound-dai = <&i2s_fifo>; + }; + + dai-link-1 { + sound-dai = <&i2s_encoder>; + dai-format = "i2s"; + mclk-fs = <256>; + + codec-0 { + sound-dai = <&codec0>; + }; + + codec-1 { + sound-dai = <&codec1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml new file mode 100644 index 000000000..580a3d040 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic T9015 Internal Audio DAC + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 0 + + compatible: + items: + - const: amlogic,t9015 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: pclk + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + AVDD-supply: + description: + Analogue power supply. + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - resets + - AVDD-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/g12a-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-reset.h> + + acodec: audio-controller@32000 { + compatible = "amlogic,t9015"; + reg = <0x32000 0x14>; + #sound-dai-cells = <0>; + clocks = <&clkc CLKID_AUDIO_CODEC>; + clock-names = "pclk"; + resets = <&reset RESET_AUDIO_CODEC>; + AVDD-supply = <&vddao_1v8>; + }; diff --git a/Documentation/devicetree/bindings/sound/apple,mca.yaml b/Documentation/devicetree/bindings/sound/apple,mca.yaml new file mode 100644 index 000000000..d5dc92b5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/apple,mca.yaml @@ -0,0 +1,131 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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> + +properties: + compatible: + items: + - enum: + - apple,t6000-mca + - apple,t8103-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' + +additionalProperties: 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 000000000..7896b8150 --- /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/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt new file mode 100644 index 000000000..953c092db --- /dev/null +++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt @@ -0,0 +1,26 @@ +Device Tree bindings for the Armada 370 DB audio +================================================ + +These Device Tree bindings are used to describe the audio complex +found on the Armada 370 DB platform. + +Mandatory properties: + + * compatible: must be "marvell,a370db-audio" + + * marvell,audio-controller: a phandle that points to the audio + controller of the Armada 370 SoC. + + * marvell,audio-codec: a set of three phandles that points to: + + 1/ the analog audio codec connected to the Armada 370 SoC + 2/ the S/PDIF transceiver + 3/ the S/PDIF receiver + +Example: + + sound { + compatible = "marvell,a370db-audio"; + marvell,audio-controller = <&audio_controller>; + marvell,audio-codec = <&audio_codec &spdif_out &spdif_in>; + }; 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 000000000..43d04702a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml @@ -0,0 +1,100 @@ +# 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: + 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: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 5 + - 10 + - 15 + - 20 + default: 10 + description: + Set non-overlapping time, the unit is nanosecond(ns). + Non-overlapping will be disabled if not specified. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clock-names + - clocks + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + classd: sound@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>; + clock-names = "pclk", "gclk"; + assigned-clocks = <&classd_gclk>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml new file mode 100644 index 000000000..0cd1ff89b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$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 000000000..f320b561f --- /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-at91sam9g20ek-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt new file mode 100644 index 000000000..9c5a9947b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt @@ -0,0 +1,26 @@ +* Atmel at91sam9g20ek wm8731 audio complex + +Required properties: + - compatible: "atmel,at91sam9g20ek-wm8731-audio" + - atmel,model: The user-visible name of this sound complex. + - atmel,audio-routing: A list of the connections between audio components. + - atmel,ssc-controller: The phandle of the SSC controller + - atmel,audio-codec: The phandle of the WM8731 audio codec +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound { + compatible = "atmel,at91sam9g20ek-wm8731-audio"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8731 @ AT91SAMG20EK"; + + atmel,audio-routing = + "Ext Spk", "LHPOUT", + "Int MIC", "MICIN"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; +}; diff --git a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt new file mode 100644 index 000000000..8facbce53 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt @@ -0,0 +1,35 @@ +* 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/wlf,wm8731.yaml + +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 new file mode 100644 index 000000000..8bbe50c88 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt @@ -0,0 +1,55 @@ +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/atmel_ac97c.txt b/Documentation/devicetree/bindings/sound/atmel_ac97c.txt new file mode 100644 index 000000000..b151bd902 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel_ac97c.txt @@ -0,0 +1,20 @@ +* Atmel AC97 controller + +Required properties: + - compatible: "atmel,at91sam9263-ac97c" + - reg: Address and length of the register set for the device + - interrupts: Should contain AC97 interrupt + - ac97-gpios: Please refer to soc-ac97link.txt, only ac97-reset is used +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound@fffa0000 { + compatible = "atmel,at91sam9263-ac97c"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ac97>; + reg = <0xfffa0000 0x4000>; + interrupts = <18 IRQ_TYPE_LEVEL_HIGH 5>; + + ac97-gpios = <&pioB 0 0 &pioB 2 0 &pioC 29 GPIO_ACTIVE_LOW>; +}; 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 000000000..274092ef3 --- /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: + - $ref: /schemas/sound/audio-graph.yaml# + +properties: + compatible: + enum: + - audio-graph-card + - audio-graph-scu-card + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "audio-graph-card"; + + dais = <&cpu_port_a>; + }; + + cpu { + /* + * dai-controller own settings + */ + + port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + dai-format = "left_j"; + }; + }; + }; + + codec { + /* + * codec own settings + */ + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml new file mode 100644 index 000000000..3de7b3682 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-card2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card2 + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - audio-graph-card2 + links: + $ref: /schemas/types.yaml#/definitions/phandle-array + label: + maxItems: 1 + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + multi: + type: object + description: Multi-CPU/Codec node + dpcm: + type: object + description: DPCM node + codec2codec: + type: object + description: Codec to Codec node + +required: + - compatible + - links + +additionalProperties: false + +examples: + - | + sound { + compatible = "audio-graph-card2"; + + links = <&cpu_port>; + }; + + cpu { + compatible = "cpu-driver"; + + cpu_port: port { cpu_ep: endpoint { remote-endpoint = <&codec_ep>; }; }; + }; + + codec { + compatible = "codec-driver"; + + port { codec_ep: endpoint { remote-endpoint = <&cpu_ep>; }; }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml new file mode 100644 index 000000000..64654ceef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-port.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card 'port' Node Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +allOf: + - $ref: /schemas/graph.yaml#/$defs/port-base + +properties: + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + convert-rate: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" + convert-channels: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + +patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: /schemas/graph.yaml#/$defs/endpoint-base + unevaluatedProperties: false + + properties: + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are + ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + frame-master: + description: Indicates dai-link frame master. + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + bitclock-master: + description: Indicates dai-link bit clock master + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + + dai-format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + convert-rate: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" + convert-channels: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + + 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. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - + description: Sample width in bits + minimum: 8 + maximum: 64 + - + description: Slot width in bits + minimum: 8 + maximum: 256 + - + description: Slot count + minimum: 1 + maximum: 64 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml new file mode 100644 index 000000000..d59baedee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -0,0 +1,46 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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> + +properties: + dais: + $ref: /schemas/types.yaml#/definitions/phandle-array + label: + maxItems: 1 + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + convert-rate: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" + convert-channels: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + + pa-gpios: + maxItems: 1 + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + +required: + - dais + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml new file mode 100644 index 000000000..dce86dafe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw8738.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic AW8738 Audio Amplifier + +maintainers: + - Stephan Gerhold <stephan@gerhold.net> + +description: + The Awinic AW8738 is a simple audio amplifier with different operation modes + (set using one-wire pulse control). The mode configures the speaker-guard + function (primarily the power limit for the amplifier). + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: awinic,aw8738 + + mode-gpios: + description: + GPIO used for one-wire pulse control. The pin is typically called SHDN + (active-low), but this is misleading since it is actually more than + just a simple shutdown/enable control. + maxItems: 1 + + awinic,mode: + description: Operation mode (number of pulses for one-wire pulse control) + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 7 + + sound-name-prefix: true + +required: + - compatible + - mode-gpios + - awinic,mode + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + audio-amplifier { + compatible = "awinic,aw8738"; + mode-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + awinic,mode = <5>; + sound-name-prefix = "Speaker Amp"; + }; diff --git a/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt new file mode 100644 index 000000000..9d049d4bf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt @@ -0,0 +1,92 @@ +Devicetree bindings for the Axentia TSE-850 audio complex + +Required properties: + - compatible: "axentia,tse850-pcm5142" + - axentia,cpu-dai: The phandle of the cpu dai. + - axentia,audio-codec: The phandle of the PCM5142 codec. + - axentia,add-gpios: gpio specifier that controls the mixer. + - axentia,loop1-gpios: gpio specifier that controls loop relays on channel 1. + - axentia,loop2-gpios: gpio specifier that controls loop relays on channel 2. + - axentia,ana-supply: Regulator that supplies the output amplifier. Must + support voltages in the 2V - 20V range, in 1V steps. + +The schematics explaining the gpios are as follows: + + loop1 relays + IN1 +---o +------------+ o---+ OUT1 + \ / + + + + | / | + +--o +--. | + | add | | + | V | + | .---. | + DAC +----------->|Sum|---+ + | '---' | + | | + + + + + IN2 +---o--+------------+--o---+ OUT2 + loop2 relays + +The 'loop1' gpio pin controlls 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'. + +In the above, the 'loop1' relays are inactive, thus feeding IN1 to the mixer +(if 'add' is active) and feeding the mixer output to OUT1. The 'loop2' relays +are active, short-cutting the TSE-850 from channel 2. IN1, IN2, OUT1 and OUT2 +are TSE-850 connectors and DAC is the PCB name of the (filtered) output from +the PCM5142 codec. + +Example: + + &ssc0 { + #sound-dai-cells = <0>; + + }; + + &i2c { + codec: pcm5142@4c { + compatible = "ti,pcm5142"; + + reg = <0x4c>; + + AVDD-supply = <®_3v3>; + DVDD-supply = <®_3v3>; + CPVDD-supply = <®_3v3>; + + clocks = <&sck>; + + pll-in = <3>; + pll-out = <6>; + }; + }; + + ana: ana-reg { + compatible = "pwm-regulator"; + + regulator-name = "ANA"; + + pwms = <&pwm0 2 1000 PWM_POLARITY_INVERTED>; + pwm-dutycycle-unit = <1000>; + pwm-dutycycle-range = <100 1000>; + + regulator-min-microvolt = <2000000>; + regulator-max-microvolt = <20000000>; + regulator-ramp-delay = <1000>; + }; + + sound { + compatible = "axentia,tse850-pcm5142"; + + axentia,cpu-dai = <&ssc0>; + axentia,audio-codec = <&codec>; + + axentia,add-gpios = <&pioA 8 GPIO_ACTIVE_LOW>; + axentia,loop1-gpios = <&pioA 10 GPIO_ACTIVE_LOW>; + axentia,loop2-gpios = <&pioA 11 GPIO_ACTIVE_LOW>; + + axentia,ana-supply = <&ana>; + }; diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt new file mode 100644 index 000000000..7bb036282 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt @@ -0,0 +1,24 @@ +* Broadcom BCM2835 SoC I2S/PCM module + +Required properties: +- compatible: "brcm,bcm2835-i2s" +- reg: Should contain PCM registers location and length. +- clocks: the (PCM) clock to use +- 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"). + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = <0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt new file mode 100644 index 000000000..007f524b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt @@ -0,0 +1,29 @@ +Broadcom DSL/PON BCM63xx Audio I2S controller + +Required properties: +- compatible: Should be "brcm,bcm63xx-i2s". +- #address-cells: 32bit valued, 1 cell. +- #size-cells: 32bit valued, 0 cell. +- reg: Should contain audio registers location and length +- interrupts: Should contain the interrupt for the controller. +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: One of each entry matching the clocks phandles list: + - "i2sclk" (generated clock) Required. + - "i2sosc" (fixed 200MHz clock) Required. + +(1) : The generated clock is required only when any of TX and RX + works on Master Mode. +(2) : The fixed 200MHz clock is from internal chip and always on + +Example: + + i2s: bcm63xx-i2s { + #address-cells = <1>; + #size-cells = <0>; + compatible = "brcm,bcm63xx-i2s"; + reg = <0xFF802080 0xFF>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&i2sclk>, <&osc>; + clock-names = "i2sclk","i2sosc"; + }; diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt new file mode 100644 index 000000000..630bf7c03 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt @@ -0,0 +1,63 @@ +BROADCOM Cygnus Audio I2S/TDM/SPDIF controller + +Required properties: + - compatible : "brcm,cygnus-audio" + - #address-cells: 32bit valued, 1 cell. + - #size-cells: 32bit valued, 0 cell. + - reg : Should contain audio registers location and length + - reg-names: names of the registers listed in "reg" property + Valid names are "aud" and "i2s_in". "aud" contains a + set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains + a set of I2S_IN registers. + - clocks: PLL and leaf clocks used by audio ports + - assigned-clocks: PLL and leaf clocks + - assigned-clock-parents: parent clocks of the assigned clocks + (usually the PLL) + - assigned-clock-rates: List of clock frequencies of the + assigned clocks + - clock-names: names of 3 leaf clocks used by audio ports + Valid names are "ch0_audio", "ch1_audio", "ch2_audio" + - interrupts: audio DMA interrupt number + +SSP Subnode properties: +- reg: The index of ssp port interface to use + Valid value are 0, 1, 2, or 3 (for spdif) + +Example: + cygnus_audio: audio@180ae000 { + compatible = "brcm,cygnus-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>; + reg-names = "aud", "i2s_in"; + clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>; + assigned-clock-rates = <1769470191>, + <0>, + <0>, + <0>; + clock-names = "ch0_audio", "ch1_audio", "ch2_audio"; + interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>; + + ssp0: ssp_port@0 { + reg = <0>; + }; + + ssp1: ssp_port@1 { + reg = <1>; + }; + + ssp2: ssp_port@2 { + reg = <2>; + }; + + spdif: spdif_port@3 { + reg = <3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 000000000..860fc0da3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml new file mode 100644 index 000000000..51d815d0c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -0,0 +1,195 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs35l41.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS35L41 Speaker Amplifier + +maintainers: + - david.rhodes@cirrus.com + +description: | + CS35L41 is a boosted mono Class D amplifier with DSP + speaker protection and equalization + +properties: + compatible: + enum: + - cirrus,cs35l40 + - cirrus,cs35l41 + + reg: + maxItems: 1 + + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + reset-gpios: + maxItems: 1 + + VA-supply: + description: voltage regulator phandle for the VA supply + + VP-supply: + description: voltage regulator phandle for the VP supply + + cirrus,boost-peak-milliamp: + description: + Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600 mA and goes to a maximum of 4500 mA with increments + of 50 mA. See section 4.3.6 of the datasheet for details. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 1600 + maximum: 4500 + default: 4500 + + cirrus,boost-ind-nanohenry: + description: + Boost inductor value, expressed in nH. Valid + values include 1000, 1200, 1500 and 2200. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 1000 + maximum: 2200 + + cirrus,boost-cap-microfarad: + description: + Total equivalent boost capacitance on the VBST + and VAMP pins, derated at 11 volts DC. The value must be rounded to the + nearest integer and expressed in uF. + $ref: "/schemas/types.yaml#/definitions/uint32" + + cirrus,asp-sdout-hiz: + description: + Audio serial port SDOUT Hi-Z control. Sets the Hi-Z + configuration for SDOUT pin of amplifier. + 0 = Logic 0 during unused slots, and while all transmit channels disabled + 1 = Hi-Z during unused slots but logic 0 while all transmit channels disabled + 2 = (Default) Logic 0 during unused slots, but Hi-Z while all transmit channels disabled + 3 = Hi-Z during unused slots and while all transmit channels disabled + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + default: 2 + + cirrus,boost-type: + description: + Configures the type of Boost being used. + Internal boost requires boost-peak-milliamp, boost-ind-nanohenry and + boost-cap-microfarad. + External Boost must have GPIO1 as GPIO output. GPIO1 will be set high to + enable boost voltage. + 0 = Internal Boost + 1 = External Boost + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 1 + + cirrus,gpio1-polarity-invert: + description: + Boolean which specifies whether the GPIO1 + level is inverted. If this property is not present the level is not inverted. + type: boolean + + cirrus,gpio1-output-enable: + description: + Boolean which specifies whether the GPIO1 pin + is configured as an output. If this property is not present the + pin will be configured as an input. + type: boolean + + cirrus,gpio1-src-select: + description: + Configures the function of the GPIO1 pin. + Note that the options are different from the GPIO2 pin + 0 = High Impedance (Default) + 1 = GPIO + 2 = Sync + 3 = MCLK input + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + + cirrus,gpio2-polarity-invert: + description: + Boolean which specifies whether the GPIO2 + level is inverted. If this property is not present the level is not inverted. + type: boolean + + cirrus,gpio2-output-enable: + description: + Boolean which specifies whether the GPIO2 pin + is configured as an output. If this property is not present the + pin will be configured as an input. + type: boolean + + cirrus,gpio2-src-select: + description: + Configures the function of the GPIO2 pin. + Note that the options are different from the GPIO1 pin. + 0 = High Impedance (Default) + 1 = GPIO + 2 = Open Drain INTB + 3 = MCLK input + 4 = Push-pull INTB (active low) + 5 = Push-pull INT (active high) + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 5 + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - if: + properties: + cirrus,boost-type: + const: 0 + then: + required: + - cirrus,boost-peak-milliamp + - cirrus,boost-ind-nanohenry + - cirrus,boost-cap-microfarad + else: + if: + properties: + cirrus,boost-type: + const: 1 + then: + required: + - cirrus,gpio1-output-enable + - cirrus,gpio1-src-select + properties: + cirrus,boost-peak-milliamp: false + cirrus,boost-ind-nanohenry: false + cirrus,boost-cap-microfarad: false + cirrus,gpio1-src-select: + enum: [1] + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l41: cs35l41@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + + cirrus,boost-type = <0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml new file mode 100644 index 000000000..184a1344e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs35l45.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS35L45 Speaker Amplifier + +maintainers: + - Ricardo Rivera-Matos <rriveram@opensource.cirrus.com> + - Richard Fitzgerald <rf@opensource.cirrus.com> + +description: | + CS35L45 is a Boosted Mono Class D Amplifier with DSP + Speaker Protection and Adaptive Battery Management. + +properties: + compatible: + enum: + - cirrus,cs35l45 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + reset-gpios: + maxItems: 1 + + vdd-a-supply: + description: voltage regulator phandle for the VDD_A supply + + vdd-batt-supply: + description: voltage regulator phandle for the VDD_BATT supply + + spi-max-frequency: + maximum: 5000000 + + cirrus,asp-sdout-hiz-ctrl: + description: + Audio serial port SDOUT Hi-Z control. Sets the Hi-Z + configuration for SDOUT pin of amplifier. Logical OR of + CS35L45_ASP_TX_HIZ_xxx values. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + default: 2 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs35l45.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l45: cs35l45@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l45"; + reg = <2>; + spi-max-frequency = <5000000>; + vdd-a-supply = <&dummy_vreg>; + vdd-batt-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + cirrus,asp-sdout-hiz-ctrl = <(CS35L45_ASP_TX_HIZ_UNUSED | + CS35L45_ASP_TX_HIZ_DISABLED)>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml new file mode 100644 index 000000000..156560b2a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$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 000000000..7356084a2 --- /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. + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 3 + + cirrus,hs-bias-sense-disable: + description: | + If present the HSBIAS sense is disabled. Configures HSBIAS output + current sense through the external 2.21-k resistor. HSBIAS_SENSE + is a hardware feature to reduce the potential pop noise when the + headset plug is removed slowly. But on some platforms ESD voltage + will affect it causing plug detection to fail, especially with CTIA + headset type. For different hardware setups, a designer might want + to tweak default behavior. + type: boolean + +required: + - compatible + - reg + - VP-supply + - VCP-supply + - VD_FILT-supply + - VL-supply + - VA-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs42l42.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l42: cs42l42@48 { + compatible = "cirrus,cs42l42"; + reg = <0x48>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD_FILT-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + + reset-gpios = <&axi_gpio_0 1 0>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + + cirrus,ts-inv = <CS42L42_TS_INV_DIS>; + cirrus,ts-dbnc-rise = <CS42L42_TS_DBNCE_1000>; + cirrus,ts-dbnc-fall = <CS42L42_TS_DBNCE_0>; + cirrus,btn-det-init-dbnce = <100>; + cirrus,btn-det-event-dbnce = <10>; + cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>; + cirrus,hs-bias-ramp-rate = <CS42L42_HSBIAS_RAMP_SLOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml new file mode 100644 index 000000000..963a871e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l51.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: CS42L51 audio codec DT bindings + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +properties: + compatible: + const: cirrus,cs42l51 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: MCLK + + reset-gpios: + maxItems: 1 + + VL-supply: + description: phandle to voltage regulator of digital interface section + + VD-supply: + description: phandle to voltage regulator of digital internal section + + VA-supply: + description: phandle to voltage regulator of analog internal section + + VAHP-supply: + description: phandle to voltage regulator of headphone + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l51@4a { + compatible = "cirrus,cs42l51"; + reg = <0x4a>; + #sound-dai-cells = <0>; + clocks = <&mclk_prov>; + clock-names = "MCLK"; + VL-supply = <®_audio>; + VD-supply = <®_audio>; + VA-supply = <®_audio>; + VAHP-supply = <®_audio>; + reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml new file mode 100644 index 000000000..cea612d3d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,lochnagar.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic Lochnagar Audio Development Board + +maintainers: + - patches@opensource.cirrus.com + +description: | + Lochnagar is an evaluation and development board for Cirrus Logic + Smart CODEC and Amp devices. It allows the connection of most Cirrus + Logic devices on mini-cards, as well as allowing connection of various + application processor systems to provide a full evaluation platform. + Audio system topology, clocking and power can all be controlled through + the Lochnagar, allowing the device under test to be used in a variety of + possible use cases. + + This binding document describes the binding for the audio portion of the + driver. + + This binding must be part of the Lochnagar MFD binding: + [1] ../mfd/cirrus,lochnagar.yaml + +properties: + compatible: + enum: + - cirrus,lochnagar2-soundcard + + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + clocks: + description: + Master clock source for the sound card, should normally be set to + LOCHNAGAR_SOUNDCARD_MCLK provided by the Lochnagar clock driver. + maxItems: 1 + + clock-names: + const: mclk + +required: + - compatible + - '#sound-dai-cells' + - clocks + - clock-names + +additionalProperties: false diff --git a/Documentation/devicetree/bindings/sound/cirrus,madera.yaml b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml new file mode 100644 index 000000000..23138ddcb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,madera.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic Madera class audio CODECs + +maintainers: + - patches@opensource.cirrus.com + +description: | + This describes audio configuration bindings for these codecs. + + See also the core bindings for the parent MFD driver: + + Documentation/devicetree/bindings/mfd/cirrus,madera.yaml + + and defines for values used in these bindings: + + include/dt-bindings/sound/madera.h + + The properties are all contained in the parent MFD node. + +properties: + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + cirrus,inmode: + description: + A list of input mode settings for each input. A maximum + of 24 cells, with four cells per input in the order INnAL, + INnAR INnBL INnBR. For non-muxed inputs the first two cells + for that input set the mode for the left and right channel + and the second two cells must be 0. For muxed inputs the + first two cells for that input set the mode of the left and + right A inputs and the second two cells set the mode of the + left and right B inputs. Valid mode values are one of the + MADERA_INMODE_xxx. If the array is shorter than the number + of inputs the unspecified inputs default to MADERA_INMODE_DIFF. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 24 + items: + minimum: 0 + maximum: 1 + default: 0 + + cirrus,out-mono: + description: + Mono bit for each output, maximum of six cells if the array + is shorter outputs will be set to stereo. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 6 + items: + minimum: 0 + maximum: 1 + default: 0 + + cirrus,dmic-ref: + description: | + Indicates how the MICBIAS pins have been externally connected + to DMICs on each input, one cell per input. + + <IN1 IN2 IN3 ...> + + A value of 0 indicates MICVDD and is the default, + other values depend on the codec: For CS47L35 one of the + CS47L35_DMIC_REF_xxx values For all other codecs one of + the MADERA_DMIC_REF_xxx values Also see the datasheet for a + description of the INn_DMIC_SUP field. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 6 + items: + minimum: 0 + maximum: 3 + default: 0 + + cirrus,max-channels-clocked: + description: + Maximum number of channels that I2S clocks will be generated + for. Useful when clock master for systems where the I2S bus + has multiple data lines. One cell for each AIF, use a value + of zero for AIFs that should be handled normally. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + default: 0 + + cirrus,pdm-fmt: + description: + PDM speaker data format, must contain 2 cells (OUT5 and + OUT6). See the PDM_SPKn_FMT field in the datasheet for a + description of this value. The second cell is ignored for + codecs that do not have OUT6. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + cirrus,pdm-mute: + description: | + PDM mute format, must contain 2 cells (OUT5 and OUT6). See the + PDM_SPKn_CTRL_1 register in the datasheet for a description + of this value. The second cell is ignored for codecs that + do not have OUT6. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/cs35l32.txt b/Documentation/devicetree/bindings/sound/cs35l32.txt new file mode 100644 index 000000000..1417d3f5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l32.txt @@ -0,0 +1,62 @@ +CS35L32 audio CODEC + +Required properties: + + - compatible : "cirrus,cs35l32" + + - reg : the I2C address of the device for I2C. Address is determined by the level + of the AD0 pin. Level 0 is 0x40 while Level 1 is 0x41. + + - VA-supply, VP-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. + + - cirrus,boost-manager : Boost voltage control. + 0 = Automatically managed. Boost-converter output voltage is the higher + of the two: Class G or adaptive LED voltage. + 1 = Automatically managed irrespective of audio, adapting for low-power + dissipation when LEDs are ON, and operating in Fixed-Boost Bypass Mode + if LEDs are OFF (VBST = VP). + 2 = (Default) Boost voltage fixed in Bypass Mode (VBST = VP). + 3 = Boost voltage fixed at 5 V. + + - cirrus,sdout-datacfg : Data configuration for dual CS35L32 applications only. + Determines the data packed in a two-CS35L32 configuration. + 0 = Left/right channels VMON[11:0], IMON[11:0], VPMON[7:0]. + 1 = Left/right channels VMON[11:0], IMON[11:0], STATUS. + 2 = (Default) left/right channels VMON[15:0], IMON [15:0]. + 3 = Left/right channels VPMON[7:0], STATUS. + + - cirrus,sdout-share : SDOUT sharing. Determines whether one or two CS35L32 + devices are on board sharing SDOUT. + 0 = (Default) One IC. + 1 = Two IC's. + + - cirrus,battery-recovery : Low battery nominal recovery threshold, rising VP. + 0 = 3.1V + 1 = 3.2V + 2 = 3.3V (Default) + 3 = 3.4V + + - cirrus,battery-threshold : Low battery nominal threshold, falling VP. + 0 = 3.1V + 1 = 3.2V + 2 = 3.3V + 3 = 3.4V (Default) + 4 = 3.5V + 5 = 3.6V + +Example: + +codec: codec@40 { + compatible = "cirrus,cs35l32"; + reg = <0x40>; + reset-gpios = <&gpio 10 0>; + cirrus,boost-manager = <0x03>; + cirrus,sdout-datacfg = <0x02>; + VA-supply = <®_audio>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l33.txt b/Documentation/devicetree/bindings/sound/cs35l33.txt new file mode 100644 index 000000000..dc5a355d1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l33.txt @@ -0,0 +1,124 @@ +CS35L33 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l33" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : gpio used to reset the amplifier + + - interrupts : IRQ line info CS35L33. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-ctl : Booster voltage use to supply the amp. If the value is + 0, then VBST = VP. If greater than 0, the boost voltage will be 3300mV with + a value of 1 and will increase at a step size of 100mV until a maximum of + 8000mV. + + - cirrus,ramp-rate : On power up, it affects the time from when the power + up sequence begins to the time the audio reaches a full-scale output. + On power down, it affects the time from when the power-down sequence + begins to when the amplifier disables the PWM outputs. If this property + is not set then soft ramping will be disabled and ramp time would be + 20ms. If this property is set to 0,1,2,3 then ramp times would be 40ms, + 60ms,100ms,175ms respectively for 48KHz sample rate. + + - cirrus,boost-ipk : The maximum current allowed for the boost converter. + The range starts at 1850000uA and goes to a maximum of 3600000uA + with a step size of 15625uA. The default is 2500000uA. + + - cirrus,imon-adc-scale : Configures the scaling of data bits from the IMON + ADC data word. This property can be set as a value of 0 for bits 15 down + to 0, 6 for 21 down to 6, 7, for 22 down to 7, 8 for 23 down to 8. + + +Optional H/G Algorithm sub-node: + +The cs35l33 node can have a single "cirrus,hg-algo" sub-node that will enable +the internal H/G Algorithm. + + - cirrus,hg-algo : Sub-node for internal Class H/G algorithm that + controls the amplifier supplies. + +Optional properties for the "cirrus,hg-algo" sub-node: + + - cirrus,mem-depth : Memory depth for the Class H/G algorithm measured in + LRCLK cycles. If this property is set to 0, 1, 2, or 3 then the memory + depths will be 1, 4, 8, 16 LRCLK cycles. The default is 16 LRCLK cycles. + + cirrus,release-rate : The number of consecutive LRCLK periods before + allowing release condition tracking updates. The number of LRCLK periods + start at 3 to a maximum of 255. + + - cirrus,ldo-thld : Configures the signal threshold at which the PWM output + stage enters LDO operation. Starts as a default value of 50mV for a value + of 1 and increases with a step size of 50mV to a maximum of 750mV (value of + 0xF). + + - cirrus,ldo-path-disable : This is a boolean property. If present, the H/G + algorithm uses the max detection path. If not present, the LDO + detection path is used. + + - cirrus,ldo-entry-delay : The LDO entry delay in milliseconds before the H/G + algorithm switches to the LDO voltage. This property can be set to values + from 0 to 7 for delays of 5ms, 10ms, 50ms, 100ms, 200ms, 500ms, 1000ms. + The default is 100ms. + + - cirrus,vp-hg-auto : This is a boolean property. When set, class H/G VPhg + automatic updating is enabled. + + - cirrus,vp-hg : Class H/G algorithm VPhg. Controls the H/G algorithm's + reference to the VP voltage for when to start generating a boosted VBST. + The reference voltage starts at 3000mV with a value of 0x3 and is increased + by 100mV per step to a maximum of 5500mV. + + - cirrus,vp-hg-rate : The rate (number of LRCLK periods) at which the VPhg is + allowed to increase to a higher voltage when using VPhg automatic + tracking. This property can be set to values from 0 to 3 with rates of 128 + periods, 2048 periods, 32768 periods, and 524288 periods. + The default is 32768 periods. + + - cirrus,vp-hg-va : VA calculation reference for automatic VPhg tracking + using VPMON. This property can be set to values from 0 to 6 starting at + 1800mV with a step size of 50mV up to a maximum value of 1750mV. + Default is 1800mV. + +Example: + +cs35l33: cs35l33@40 { + compatible = "cirrus,cs35l33"; + reg = <0x40>; + + VA-supply = <&ldo5_reg>; + VP-supply = <&ldo5_reg>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&cs47l91 34 0>; + + cirrus,ramp-rate = <0x0>; + cirrus,boost-ctl = <0x30>; /* VBST = 8000mV */ + cirrus,boost-ipk = <0xE0>; /* 3600mA */ + cirrus,imon-adc-scale = <0> /* Bits 15 down to 0 */ + + cirrus,hg-algo { + cirrus,mem-depth = <0x3>; + cirrus,release-rate = <0x3>; + cirrus,ldo-thld = <0x1>; + cirrus,ldo-path-disable = <0x0>; + cirrus,ldo-entry-delay=<0x4>; + cirrus,vp-hg-auto; + cirrus,vp-hg=<0xF>; + cirrus,vp-hg-rate=<0x2>; + cirrus,vp-hg-va=<0x0>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l34.txt b/Documentation/devicetree/bindings/sound/cs35l34.txt new file mode 100644 index 000000000..2f7606b7d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l34.txt @@ -0,0 +1,62 @@ +CS35L34 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l34" + + - reg : the I2C address of the device for I2C. + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-vtge-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from VP to 8V with + increments of 100mV. + + - cirrus,boost-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + + - reset-gpios: GPIO used to reset the amplifier. + + - interrupts : IRQ line info CS35L34. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The + range starts at 1200mA and goes to a maximum of 3840mA with increments of + 80mA. The default value is 2480mA. + + - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the + received mono data is in the left or right portion of the I2S frame + according to the AD0 pin or directly via this configuration. + 0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel), + 0x2 = Left, + 0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right + channel), + 0x3 = Right. + + - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take + effect without waiting for a zero cross. + + - cirrus,tdm-rising-edge: Boolean property. If set, data is on the rising edge of + SCLK. Otherwise, data is on the falling edge of SCLK. + + +Example: + +cs35l34: cs35l34@40 { + compatible = "cirrus,cs35l34"; + reg = <0x40>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&gpio 10 0>; + + cirrus,boost-vtge-milltvolt = <8000>; /* 8V */ + cirrus,boost-ind-nanohenry = <1000>; /* 1uH */ + cirrus,boost-peak-milliamp = <3000>; /* 3A */ +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 000000000..7915897f8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,181 @@ +CS35L35 Boosted Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-ind-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + - reset-gpios : gpio used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,external-boost : Boolean to specify the device is using an external + boost supply, note that sharing a boost from another cs35l35 would constitute + using an external supply for the slave device + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + - cirrus,sp-drv-unused : Determines how unused slots should be driven on the + Serial Port. + 0 - Hi-Z + 2 - Drive 0's (Default) + 3 - Drive 1's + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2600mV to 9000mV with + increments of 100mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1680mA and goes to a maximum of 4480mA with increments of + 110mA. + (Default) 2.46 Amps + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + 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 + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 4 8 bit values to set the depth, location, frame and ADC + scale of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: cs35l35@20 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl-millivolt = <9000>; + + cirrus,stereo-config; + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 000000000..912bd162b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0x00>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt new file mode 100644 index 000000000..380fff8e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt new file mode 100644 index 000000000..c33770ec4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -0,0 +1,21 @@ +CS4270 audio CODEC + +The driver for this device currently only supports I2C. + +Required properties: + + - compatible : "cirrus,cs4270" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Example: + +codec: cs4270@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt new file mode 100644 index 000000000..6e699ceab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -0,0 +1,57 @@ +Cirrus Logic CS4271 DT bindings + +This driver supports both the I2C and the SPI bus. + +Required properties: + + - compatible: "cirrus,cs4271" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the i2c address + + +Optional properties: + + - reset-gpio: a GPIO spec to define which pin is connected to the chip's + !RESET pin + - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag + is enabled. + - cirrus,enable-soft-reset: + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. + + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/Documentation/devicetree/bindings/sound/cs42l52.txt b/Documentation/devicetree/bindings/sound/cs42l52.txt new file mode 100644 index 000000000..bc03c9312 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l52.txt @@ -0,0 +1,46 @@ +CS42L52 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l52" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - cirrus,reset-gpio : GPIO controller's phandle and the number + of the GPIO used to reset the codec. + + - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency. + Allowable values of 0x00 through 0x0F. These are raw values written to the + register, not the actual frequency. The frequency is determined by the following. + Frequency = (64xFs)/(N+2) + N = chgfreq_val + Fs = Sample Rate (variable) + + - cirrus,mica-differential-cfg : boolean, If present, then the MICA input is configured + as a differential input. If not present then the MICA input is configured as + Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input. + + - cirrus,micb-differential-cfg : boolean, If present, then the MICB input is configured + as a differential input. If not present then the MICB input is configured as + Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input. + + - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin + 0 = 0.5 x VA + 1 = 0.6 x VA + 2 = 0.7 x VA + 3 = 0.8 x VA + 4 = 0.83 x VA + 5 = 0.91 x VA + +Example: + +codec: codec@4a { + compatible = "cirrus,cs42l52"; + reg = <0x4a>; + reset-gpio = <&gpio 10 0>; + cirrus,chgfreq-divisor = <0x05>; + cirrus.mica-differential-cfg; + cirrus,micbias-lvl = <5>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs42l56.txt b/Documentation/devicetree/bindings/sound/cs42l56.txt new file mode 100644 index 000000000..4ba520a28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l56.txt @@ -0,0 +1,63 @@ +CS42L52 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l56" + + - reg : the I2C address of the device for I2C + + - VA-supply, VCP-supply, VLDO-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - cirrus,gpio-nreset : GPIO controller's phandle and the number + of the GPIO used to reset the codec. + + - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency. + Allowable values of 0x00 through 0x0F. These are raw values written to the + register, not the actual frequency. The frequency is determined by the following. + Frequency = MCLK / 4 * (N+2) + N = chgfreq_val + MCLK = Where MCLK is the frequency of the mclk signal after the MCLKDIV2 circuit. + + - cirrus,ain1a-ref-cfg, ain1b-ref-cfg : boolean, If present, AIN1A or AIN1B are configured + as a pseudo-differential input referenced to AIN1REF/AIN3A. + + - cirrus,ain2a-ref-cfg, ain2b-ref-cfg : boolean, If present, AIN2A or AIN2B are configured + as a pseudo-differential input referenced to AIN2REF/AIN3B. + + - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin. + 0 = 0.5 x VA + 1 = 0.6 x VA + 2 = 0.7 x VA + 3 = 0.8 x VA + 4 = 0.83 x VA + 5 = 0.91 x VA + + - cirrus,adaptive-pwr-cfg : Configures how the power to the Headphone and Lineout + Amplifiers adapt to the output signal levels. + 0 = Adapt to Volume Mode. Voltage level determined by the sum of the relevant volume settings. + 1 = Fixed - Headphone and Line Amp supply = + or - VCP/2. + 2 = Fixed - Headphone and Line Amp supply = + or - VCP. + 3 = Adapted to Signal; Voltage level is dynamically determined by the output signal. + + - cirrus,hpf-left-freq, hpf-right-freq : Sets the corner frequency (-3dB point) for the internal High-Pass + Filter. + 0 = 1.8Hz + 1 = 119Hz + 2 = 236Hz + 3 = 464Hz + + +Example: + +codec: codec@4b { + compatible = "cirrus,cs42l56"; + reg = <0x4b>; + cirrus,gpio-nreset = <&gpio 10 0>; + cirrus,chgfreq-divisor = <0x05>; + cirrus.ain1_ref_cfg; + cirrus,micbias-lvl = <5>; + VA-supply = <®_audio>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt new file mode 100644 index 000000000..47b868b5a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l73.txt @@ -0,0 +1,22 @@ +CS42L73 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l73" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset_gpio : a GPIO spec for the reset pin. + - chgfreq : Charge Pump Frequency values 0x00-0x0F + + +Example: + +codec: cs42l73@4a { + compatible = "cirrus,cs42l73"; + reg = <0x4a>; + reset_gpio = <&gpio 10 0>; + chgfreq = <0x05>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt new file mode 100644 index 000000000..bbfe39347 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -0,0 +1,34 @@ +CS42448/CS42888 audio CODEC + +Required properties: + + - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + + - reset-gpios : a GPIO spec to define which pin is connected to the chip's + !RESET pin + +Example: + +cs42888: codec@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <®_audio>; + VD-supply = <®_audio>; + VLS-supply = <®_audio>; + VLC-supply = <®_audio>; + reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 000000000..8b1dd5aeb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,67 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130", "cirrus,cs4399", "cirrus,cs43131", + "cirrus,cs43198" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + + - cirrus,dc-measure: + Boolean, define to enable headphone DC impedance measurement. + + - cirrus,ac-measure: + Boolean, define to enable headphone AC impedance measurement. + DC impedance must also be enabled for AC impedance measurement. + + - cirrus,dc-threshold: + Define 2 DC impedance thresholds in ohms for HP output control. + Default values are 50 and 120 Ohms. + + - cirrus,ac-freq: + Define the frequencies at which to measure HP AC impedance. + Only used if "cirrus,dc-measure" is defined. + Exactly 10 frequencies must be defined. + If this properties is undefined, by default, + following frequencies are used: + <24 43 93 200 431 928 2000 4309 9283 20000> + The above frequencies are logarithmically equally spaced. + Log base is 10. + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + cirrus,dc-measure; + cirrus,ac-measure; + cirrus,dc-threshold = /bits/ 16 <20 100>; + cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt new file mode 100644 index 000000000..12b4aa8ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -0,0 +1,22 @@ +Cirrus Logic CS4341 audio DAC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "cirrus,cs4341a" + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +For required properties on I2C-bus, please consult +Documentation/devicetree/bindings/i2c/i2c.txt +For required properties on SPI-bus, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Example: + codec: cs4341@0 { + #sound-dai-cells = <0>; + compatible = "cirrus,cs4341a"; + reg = <0>; + spi-max-frequency = <6000000>; + }; diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 000000000..54c117b59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 000000000..4dbfb8274 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,44 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + + - VA-supply, VP-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. + + - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either + GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled + by the driver automatically. + + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. + 0 = Hi-Z + 1 = 1.80 V + 2 = 2.75 V + + - cirrus,use-sdout2 : This is a boolean property. If present, it indicates + the hardware design connects both SDOUT1 and SDOUT2 + pins to output data. Otherwise, it indicates that + only SDOUT1 is connected for data output. + * CS53l30 supports 4-channel data output in the same + * 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 + * 4-channel data per frame. + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; + VA-supply = <&cs53l30_va>; + VP-supply = <&cs53l30_vp>; +}; diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt new file mode 100644 index 000000000..94584c96c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da7213.txt @@ -0,0 +1,45 @@ +Dialog Semiconductor DA7212/DA7213 Audio Codec bindings + +====== + +Required properties: +- compatible : Should be "dlg,da7212" or "dlg,da7213" +- reg: Specifies the I2C slave address + +Optional properties: +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1 + [<1600>, <2200>, <2500>, <3000>] +- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2 + [<1600>, <2200>, <2500>, <3000>] +- dlg,dmic-data-sel : DMIC channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic-samplephase : When to sample audio from DMIC. + ["on_clkedge", "between_clkedge"] +- dlg,dmic-clkrate : DMIC clock frequency (Hz). + [<1500000>, <3000000>] + + - VDDA-supply : Regulator phandle for Analogue power supply + - VDDMIC-supply : Regulator phandle for Mic Bias + - VDDIO-supply : Regulator phandle for I/O power supply + +====== + +Example: + + codec_i2c: da7213@1a { + compatible = "dlg,da7213"; + reg = <0x1a>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias1-lvl = <2500>; + dlg,micbias2-lvl = <2500>; + + dlg,dmic-data-sel = "lrise_rfall"; + dlg,dmic-samplephase = "between_clkedge"; + dlg,dmic-clkrate = <3000000>; + }; diff --git a/Documentation/devicetree/bindings/sound/da7218.txt b/Documentation/devicetree/bindings/sound/da7218.txt new file mode 100644 index 000000000..2cf30899b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da7218.txt @@ -0,0 +1,102 @@ +Dialog Semiconductor DA7218 Audio Codec bindings + +DA7218 is an audio codec with HP detect feature. + +====== + +Required properties: +- compatible : Should be "dlg,da7217" or "dlg,da7218" +- reg: Specifies the I2C slave address + +- 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: +- interrupts: IRQ line info for DA7218 chip. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for + further information relating to interrupt 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). + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias1-lvl-millivolt : Voltage (mV) for Mic Bias 1 + [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>] +- dlg,micbias2-lvl-millivolt : Voltage (mV) for Mic Bias 2 + [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>] +- dlg,mic1-amp-in-sel : Mic1 input source type + ["diff", "se_p", "se_n"] +- dlg,mic2-amp-in-sel : Mic2 input source type + ["diff", "se_p", "se_n"] +- dlg,dmic1-data-sel : DMIC1 channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic1-samplephase : When to sample audio from DMIC1. + ["on_clkedge", "between_clkedge"] +- dlg,dmic1-clkrate-hz : DMic1 clock frequency (Hz). + [<1500000>, <3000000>] +- dlg,dmic2-data-sel : DMic2 channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic2-samplephase : When to sample audio from DMic2. + ["on_clkedge", "between_clkedge"] +- dlg,dmic2-clkrate-hz : DMic2 clock frequency (Hz). + [<1500000>, <3000000>] +- dlg,hp-diff-single-supply : Boolean flag, use single supply for HP + (DA7217 only) + +====== + +Optional Child node - 'da7218_hpldet' (DA7218 only): + +Optional properties: +- dlg,jack-rate-us : Time between jack detect measurements (us) + [<5>, <10>, <20>, <40>, <80>, <160>, <320>, <640>] +- dlg,jack-debounce : Number of debounce measurements taken for jack detect + [<0>, <2>, <3>, <4>] +- dlg,jack-threshold-pct : Threshold level for jack detection (% of VDD) + [<84>, <88>, <92>, <96>] +- dlg,comp-inv : Boolean flag, invert comparator output +- dlg,hyst : Boolean flag, enable hysteresis +- dlg,discharge : Boolean flag, auto discharge of Mic Bias on jack removal + +====== + +Example: + + codec: da7218@1a { + compatible = "dlg,da7218"; + reg = <0x1a>; + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + wakeup-source; + + VDD-supply = <®_audio>; + VDDMIC-supply = <®_audio>; + VDDIO-supply = <®_audio>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias1-lvl-millivolt = <2600>; + dlg,micbias2-lvl-millivolt = <2600>; + dlg,mic1-amp-in-sel = "diff"; + dlg,mic2-amp-in-sel = "diff"; + + dlg,dmic1-data-sel = "lrise_rfall"; + dlg,dmic1-samplephase = "on_clkedge"; + dlg,dmic1-clkrate-hz = <3000000>; + dlg,dmic2-data-sel = "lrise_rfall"; + dlg,dmic2-samplephase = "on_clkedge"; + dlg,dmic2-clkrate-hz = <3000000>; + + da7218_hpldet { + dlg,jack-rate-us = <40>; + dlg,jack-debounce = <2>; + dlg,jack-threshold-pct = <84>; + dlg,hyst; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt new file mode 100644 index 000000000..add1caf26 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da7219.txt @@ -0,0 +1,112 @@ +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 new file mode 100644 index 000000000..75c6338b6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da9055.txt @@ -0,0 +1,22 @@ +* Dialog DA9055 Audio CODEC + +DA9055 provides Audio CODEC support (I2C only). + +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: +Documentation/devicetree/bindings/mfd/da9055.txt + +Required properties: + + - compatible: "dlg,da9055-codec" + - reg: Specifies the I2C slave address + + +Example: + + codec: da9055-codec@1a { + compatible = "dlg,da9055-codec"; + reg = <0x1a>; + }; diff --git a/Documentation/devicetree/bindings/sound/dai-params.yaml b/Documentation/devicetree/bindings/sound/dai-params.yaml new file mode 100644 index 000000000..f5fb71f9b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dai-params.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/dai-params.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface (DAI) Stream Parameters + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +$defs: + + dai-channels: + description: Number of audio channels used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 32 + + dai-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 + + dai-sample-rate: + description: Audio sample rate used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 8000 + maximum: 192000 + +properties: {} + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 000000000..963e10051 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,49 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx +- ti,model : The user-visible name of this sound complex. +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,mcasp-controller : The phandle of the McASP controller +- ti,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 codec's pins, and the jacks on the board: + +Optional properties: +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec. +- clocks : Reference to the master clock +- clock-names : The clock should be named "mclk" +- Either codec-clock-rate or the codec-clock reference has to be defined. If + the both are defined the driver attempts to set referenced clock to the + defined rate and takes the rate from the clock reference. + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + * Line In + + +Example: + +sound { + compatible = "ti,da830-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + ti,audio-routing = + "Headphone Jack", "HPLOUT", + "Headphone Jack", "HPROUT", + "Line Out", "LLOUT", + "Line Out", "RLOUT", + "MIC3L", "Mic Bias 2V", + "MIC3R", "Mic Bias 2V", + "Mic Bias 2V", "Mic Jack", + "LINE1L", "Line In", + "LINE2L", "Line In", + "LINE1R", "Line In", + "LINE2R", "Line In"; +}; 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 000000000..f46c66bc6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary@ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: 0 - I2S or 1 - DIT operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number of channels over one serializer + the property is ignored in DIT mode + minimum: 2 + maximum: 32 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures WFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures RFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX interrupt + - description: RX interrupt + - items: + - description: common/combined interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +allOf: + - if: + properties: + opmode: + enum: + - 0 + + then: + required: + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt new file mode 100644 index 000000000..3ffc2562f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt @@ -0,0 +1,50 @@ +Texas Instruments DaVinci McBSP module +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This binding describes the "Multi-channel Buffered Serial Port" (McBSP) +audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x. + + +Required properties: +~~~~~~~~~~~~~~~~~~~~ +- compatible : + "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms + +- reg : physical base address and length of the controller memory mapped + region(s). +- reg-names : Should contain: + * "mpu" for the main registers (required). + * "dat" for the data FIFO (optional). + +- dmas: three element list of DMA controller phandles, DMA request line and + TC channel ordered triplets. +- 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: +~~~~~~~~~~~~~~~~~~~~ +- interrupts : Interrupt numbers for McBSP +- interrupt-names : Known interrupt names are "rx" and "tx" + +- 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 + +Example (AM1808): +~~~~~~~~~~~~~~~~~ + +mcbsp0: mcbsp@1d10000 { + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x00110000 0x1000>, + <0x00310000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97 98>; + interrupt-names = "rx", "tx"; + dmas = <&edma0 3 1 + &edma0 2 1>; + dma-names = "tx", "rx"; +}; diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt new file mode 100644 index 000000000..32e871037 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -0,0 +1,22 @@ +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 new file mode 100644 index 000000000..33fbf058c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/es8328.txt @@ -0,0 +1,38 @@ +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/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 000000000..6dfa88c4d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,26 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: + + - compatible : "eukrea,asoc-tlv320" + + - eukrea,model : The user-visible name of this sound complex. + + - ssi-controller : The phandle of the SSI controller. + + - fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). + + - fsl,mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es7134.txt b/Documentation/devicetree/bindings/sound/everest,es7134.txt new file mode 100644 index 000000000..091666069 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es7134.txt @@ -0,0 +1,15 @@ +ES7134 i2s DA converter + +Required properties: +- compatible : "everest,es7134" or + "everest,es7144" or + "everest,es7154" +- VDD-supply : regulator phandle for the VDD supply +- PVDD-supply: regulator phandle for the PVDD supply for the es7154 + +Example: + +i2s_codec: external-codec { + compatible = "everest,es7134"; + VDD-supply = <&vcc_5v>; +}; diff --git a/Documentation/devicetree/bindings/sound/everest,es7241.txt b/Documentation/devicetree/bindings/sound/everest,es7241.txt new file mode 100644 index 000000000..28f82cf49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es7241.txt @@ -0,0 +1,28 @@ +ES7241 i2s AD converter + +Required properties: +- compatible : "everest,es7241" +- VDDP-supply: regulator phandle for the VDDA supply +- VDDA-supply: regulator phandle for the VDDP supply +- VDDD-supply: regulator phandle for the VDDD supply + +Optional properties: +- reset-gpios: gpio connected to the reset pin +- m0-gpios : gpio connected to the m0 pin +- m1-gpios : gpio connected to the m1 pin +- everest,sdout-pull-down: + Format used by the serial interface is controlled by pulling + the sdout. If the sdout is pulled down, leftj format is used. + If this property is not provided, sdout is assumed to pulled + up and i2s format is used + +Example: + +linein: audio-codec@2 { + #sound-dai-cells = <0>; + compatible = "everest,es7241"; + VDDA-supply = <&vcc_3v3>; + VDDP-supply = <&vcc_3v3>; + VDDD-supply = <&vcc_3v3>; + reset-gpios = <&gpio GPIOH_42>; +}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000..3b752bba7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8316 audio CODEC + +maintainers: + - Daniel Drake <drake@endlessm.com> + - Katsuhiro Suzuki <katsuhiro@katsuster.net> + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100644 index 000000000..07781408e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + $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: 0x45 + +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,mic1-src = [22]; + everest,mic2-src = [44]; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [45]; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt new file mode 100644 index 000000000..998b4c8a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -0,0 +1,80 @@ +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 000000000..aa4be7170 --- /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,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt new file mode 100644 index 000000000..840b7e0d6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt @@ -0,0 +1,50 @@ +NXP Audio Mixer (AUDMIX). + +The Audio Mixer is a on-chip functional module that allows mixing of two +audio streams into a single audio stream. Audio Mixer has two input serial +audio interfaces. These are driven by two Synchronous Audio interface +modules (SAI). Each input serial interface carries 8 audio channels in its +frame in TDM manner. Mixer mixes audio samples of corresponding channels +from two interfaces into a single sample. Before mixing, audio samples of +two inputs can be attenuated based on configuration. The output of the +Audio Mixer is also a serial audio interface. Like input interfaces it has +the same TDM frame format. This output is used to drive the serial DAC TDM +interface of audio codec and also sent to the external pins along with the +receive path of normal audio SAI module for readback by the CPU. + +The output of Audio Mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - mixed audio + +Mixing operation is independent of audio sample rate but the two audio +input streams must have same audio sample rate with same number of channels +in TDM frame to be eligible for mixing. + +Device driver required properties: +================================= + - compatible : Compatible list, contains "fsl,imx8qm-audmix" + + - 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 "ipg" for register access. + + - power-domains : Must contain the phandle to AUDMIX power domain node + + - dais : Must contain a list of phandles to AUDMIX connected + DAIs. The current implementation requires two phandles + to SAI interfaces to be provided, the first SAI in the + list being used to route the AUDMIX output. + +Device driver configuration example: +====================================== + audmix: audmix@59840000 { + compatible = "fsl,imx8qm-audmix"; + reg = <0x0 0x59840000 0x0 0x10000>; + clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>; + clock-names = "ipg"; + power-domains = <&pd_audmix>; + dais = <&sai4>, <&sai5>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml new file mode 100644 index 000000000..bdde68a10 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,easrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Asynchronous Sample Rate Converter (ASRC) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + $nodename: + pattern: "^easrc@.*" + + compatible: + const: fsl,imx8mn-easrc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: mem + + dmas: + maxItems: 8 + + dma-names: + items: + - const: ctx0_rx + - const: ctx0_tx + - const: ctx1_rx + - const: ctx1_tx + - const: ctx2_rx + - const: ctx2_tx + - const: ctx3_rx + - const: ctx3_tx + + firmware-name: + $ref: /schemas/types.yaml#/definitions/string + const: imx/easrc/easrc-imx8mn.bin + description: The coefficient table for the filters + + fsl,asrc-rate: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 8000 + maximum: 192000 + description: Defines a mutual sample rate used by DPCM Back Ends + + fsl,asrc-format: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [2, 6, 10, 32, 36] + default: 2 + description: + Defines a mutual sample format used by DPCM Back Ends + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - firmware-name + - fsl,asrc-rate + - fsl,asrc-format + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + easrc: easrc@300c0000 { + compatible = "fsl,imx8mn-easrc"; + reg = <0x300c0000 0x10000>; + interrupts = <0x0 122 0x4>; + clocks = <&clk IMX8MN_CLK_ASRC_ROOT>; + clock-names = "mem"; + dmas = <&sdma2 16 23 0> , <&sdma2 17 23 0>, + <&sdma2 18 23 0> , <&sdma2 19 23 0>, + <&sdma2 20 23 0> , <&sdma2 21 23 0>, + <&sdma2 22 23 0> , <&sdma2 23 23 0>; + dma-names = "ctx0_rx", "ctx0_tx", + "ctx1_rx", "ctx1_tx", + "ctx2_rx", "ctx2_tx", + "ctx3_rx", "ctx3_tx"; + firmware-name = "imx/easrc/easrc-imx8mn.bin"; + fsl,asrc-rate = <8000>; + fsl,asrc-format = <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt new file mode 100644 index 000000000..0a2480aee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -0,0 +1,68 @@ +Freescale Enhanced Serial Audio Interface (ESAI) Controller + +The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port +for serial communication with a variety of serial devices, including industry +standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and +other DSPs. It has up to six transmitters and four receivers. + +Required properties: + + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-esai", + "fsl,vf610-esai", + "fsl,imx6ull-esai", + "fsl,imx8qm-esai", + + - reg : Offset and length of the register set for the device. + + - 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 used to access registers + "extal" The esai baud clock for esai controller used to + derive HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to + derive HCK, SCK and FS. + "spba" The spba clock is required when ESAI 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,fifo-depth : The number of elements in the transmit and receive + FIFOs. This number is the maximum allowed value for + TFCR[TFWM] or RFCR[RFWM]. + + - 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. + +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: + +esai: esai@2024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + big-endian; +}; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml new file mode 100644 index 000000000..64d57758e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.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/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: + enum: + - fsl,imx8mm-micfil + - fsl,imx8mp-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 + +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 new file mode 100644 index 000000000..d66284b8b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,36 @@ +fsl,mqs audio CODEC + +Required properties: + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs", "fsl,imx93-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : "mclk" - must required. + "core" - required if compatible is "fsl,imx8qm-mqs", it + is for register access. + - gpr : A phandle of General Purpose Registers in IOMUX Controller. + Required if compatible is "fsl,imx6sx-mqs". + +Required if compatible is "fsl,imx8qm-mqs": + - power-domains: A phandle of PM domain provider node. + - reg: Offset and length of the register set for the device. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; + +mqs: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&clk IMX8QM_AUD_MQS_IPG>, + <&clk IMX8QM_AUD_MQS_HMCLK>; + clock-names = "core", "mclk"; + power-domains = <&pd_mqs0>; + status = "disabled"; +}; diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml new file mode 100644 index 000000000..d370c98a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio RPMSG CPU DAI Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + fsl_rpmsg is a virtual audio device. Mapping to real hardware devices + are SAI, DMA controlled by Cortex M core. What we see from Linux + side is a device which provides audio service by rpmsg channel. + +properties: + compatible: + enum: + - fsl,imx7ulp-rpmsg-audio + - fsl,imx8mn-rpmsg-audio + - fsl,imx8mm-rpmsg-audio + - fsl,imx8mp-rpmsg-audio + - fsl,imx8ulp-rpmsg-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + clocks: + items: + - description: Peripheral clock for register access + - description: Master clock + - description: DMA clock for DMA register access + - description: Parent clock for multiple of 8kHz sample rates + - description: Parent clock for multiple of 11kHz sample rates + + clock-names: + items: + - const: ipg + - const: mclk + - const: dma + - const: pll8k + - const: pll11k + + power-domains: + description: + List of phandle and PM domain specifier as documented in + Documentation/devicetree/bindings/power/power_domain.txt + maxItems: 1 + + memory-region: + maxItems: 1 + description: + phandle to a node describing reserved memory (System RAM memory) + The M core can't access all the DDR memory space on some platform, + So reserved a specific memory for dma buffer which M core can + access. + (see bindings/reserved-memory/reserved-memory.txt) + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to a node of audio codec + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + fsl,enable-lpa: + $ref: /schemas/types.yaml#/definitions/flag + description: enable low power audio path. + + fsl,rpmsg-out: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, the transmitting function + will be enabled. + + fsl,rpmsg-in: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, the receiving function + will be enabled. + +required: + - compatible + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + rpmsg_audio: rpmsg_audio { + compatible = "fsl,imx8mn-rpmsg-audio"; + model = "wm8524-audio"; + fsl,enable-lpa; + fsl,rpmsg-out; + clocks = <&clk IMX8MN_CLK_SAI3_IPG>, + <&clk IMX8MN_CLK_SAI3_ROOT>, + <&clk IMX8MN_CLK_SDMA3_ROOT>, + <&clk IMX8MN_AUDIO_PLL1_OUT>, + <&clk IMX8MN_AUDIO_PLL2_OUT>; + clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000..70c4111d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,216 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + - enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + 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: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + 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 + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + 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,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,shared-interrupt: + description: Interrupt is shared with other modules. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + 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 + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +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/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 = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + 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.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 000000000..1d64e8337 --- /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,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt new file mode 100644 index 000000000..7e15a85ce --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -0,0 +1,87 @@ +Freescale Synchronous Serial Interface + +The SSI is a serial device that communicates with audio codecs. It can +be programmed in AC97, I2S, left-justified, or right-justified modes. + +Required properties: +- compatible: Compatible list, should contain one of the following + compatibles: + fsl,mpc8610-ssi + fsl,imx51-ssi + fsl,imx35-ssi + fsl,imx21-ssi +- cell-index: The SSI, <0> = SSI1, <1> = SSI2, and so on. +- reg: Offset and length of the register set for the device. +- interrupts: <a b> where a is the interrupt number and b is a + field that represents an encoding of the sense and + level information for the interrupt. This should be + encoded based on the information in section 2) + depending on the type of interrupt controller you + have. +- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for SFCSR[TFWM0]. + - clocks: "ipg" - Required clock for the SSI unit + "baud" - Required clock for SSI master mode. Otherwise this + clock is not used + +Required are also ac97 link bindings if ac97 is used. See +Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary +bindings. + +Optional properties: +- codec-handle: Phandle to a 'codec' node that defines an audio + codec connected to this SSI. This node is typically + a child of an I2C or other control node. +- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to + filter the codec stream. This is necessary for some boards + where an incompatible codec is connected to this SSI, e.g. + on pca100 and pcm043. +- 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", if fsl,imx-fiq + is not defined. +- fsl,mode: The operating mode for the AC97 interface only. + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master +- fsl,ssi-asynchronous: + If specified, the SSI is to be programmed in asynchronous + mode. In this mode, pins SRCK, STCK, SRFS, and STFS must + all be connected to valid signals. In synchronous mode, + SRCK and SRFS are ignored. Asynchronous mode allows + playback and capture to use different sample sizes and + sample rates. Some drivers may require that SRCK and STCK + be connected together, and SRFS and STFS be connected + together. This would still allow different sample sizes, + but not different sample rates. +- fsl,playback-dma: Phandle to a node for the DMA channel to use for + playback of audio. This is typically dictated by SOC + design. See the notes below. + Only used on Power Architecture. +- fsl,capture-dma: Phandle to a node for the DMA channel to use for + capture (recording) of audio. This is typically dictated + by SOC design. See the notes below. + Only used on Power Architecture. + +Child 'codec' node required properties: +- compatible: Compatible list, contains the name of the codec + +Child 'codec' node optional properties: +- clock-frequency: The frequency of the input clock, which typically comes + from an on-board dedicated oscillator. + +Notes on fsl,playback-dma and fsl,capture-dma: + +On SOCs that have an SSI, specific DMA channels are hard-wired for playback +and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for +playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for +playback and DMA channel 3 for capture. The developer can choose which +DMA controller to use, but the channels themselves are hard-wired. The +purpose of these two properties is to represent this hardware design. + +The device tree nodes for the DMA channels that are referenced by +"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with +"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g. +"fsl,mpc8610-dma-channel") can remain. If these nodes are left as +"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA +drivers (fsldma) will attempt to use them, and it will conflict with the +sound drivers. diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml new file mode 100644 index 000000000..223b8ea69 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml @@ -0,0 +1,104 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,xcvr.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Transceiver (XCVR) Controller + +maintainers: + - Viorel Suman <viorel.suman@nxp.com> + +description: | + NXP XCVR (Audio Transceiver) is a on-chip functional module + that allows CPU to receive and transmit digital audio via + HDMI2.1 eARC, HDMI1.4 ARC and SPDIF. + +properties: + $nodename: + pattern: "^xcvr@.*" + + compatible: + enum: + - fsl,imx8mp-xcvr + + reg: + items: + - description: 20K RAM for code and data + - description: registers space + - description: RX FIFO address + - description: TX FIFO address + + reg-names: + items: + - const: ram + - const: regs + - const: rxfifo + - const: txfifo + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + - description: PHY clock + - description: SPBA clock + - description: PLL clock + + clock-names: + items: + - const: ipg + - const: phy + - const: spba + - const: pll_ipg + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + +required: + - compatible + - reg + - reg-names + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + #include <dt-bindings/reset/imx8mp-reset.h> + + xcvr: xcvr@30cc0000 { + compatible = "fsl,imx8mp-xcvr"; + reg = <0x30cc0000 0x800>, + <0x30cc0800 0x400>, + <0x30cc0c00 0x080>, + <0x30cc0e00 0x080>; + reg-names = "ram", "regs", "rxfifo", "txfifo"; + interrupts = <0x0 128 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_IPG>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_PHY>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_SPBA2_ROOT>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUDPLL_ROOT>; + clock-names = "ipg", "phy", "spba", "pll_ipg"; + dmas = <&sdma2 30 2 0>, <&sdma2 31 2 0>; + dma-names = "rx", "tx"; + resets = <&audiomix_reset 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt new file mode 100644 index 000000000..8b4f4015c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -0,0 +1,115 @@ +Freescale Generic ASoC Sound Card with ASRC support + +The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale +SoCs connecting with external CODECs. + +The idea of this generic sound card is a bit like ASoC Simple Card. However, +for Freescale SoCs (especially those released in recent years), most of them +have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And +this is a specific feature that might be painstakingly controlled and merged +into the Simple Card. + +So having this generic sound card allows all Freescale SoC users to benefit +from the simplification of a new card support and the capability of the wide +sample rates support through ASRC. + +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. + + +The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + + "fsl,imx-audio-cs42888" + + "fsl,imx-audio-cs427x" + (compatible with CS4271 and CS4272) + + "fsl,imx-audio-wm8962" + + "fsl,imx-audio-sgtl5000" + (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) + + "fsl,imx-audio-wm8960" + + "fsl,imx-audio-mqs" + + "fsl,imx-audio-wm8524" + + "fsl,imx-audio-tlv320aic32x4" + + "fsl,imx-audio-tlv320aic31xx" + + "fsl,imx-audio-si476x" + + "fsl,imx-audio-wm8958" + +Required properties: + + - compatible : Contains one of entries in the compatible list. + + - model : The user-visible name of this sound complex + + - audio-cpu : The phandle of an CPU DAI controller + + - 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 + source. There're a few pre-designed board connectors: + * Line Out Jack + * Line In Jack + * Headphone Jack + * Mic Jack + * Ext Spk + * AMIC (stands for Analog Microphone Jack) + * DMIC (stands for Digital Microphone Jack) + + Note: The "Mic Jack" and "AMIC" are redundant while + coexisting in order to support the old bindings + of wm8962 and sgtl5000. + + - 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: + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer + +Example: +sound-cs42888 { + compatible = "fsl,imx-audio-cs42888"; + model = "cs42888-audio"; + audio-cpu = <&esai>; + audio-asrc = <&asrc>; + audio-codec = <&cs42888>; + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "Line In Jack", + "AIN1R", "Line In Jack", + "AIN2L", "Line In Jack", + "AIN2R", "Line In Jack"; +}; diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml new file mode 100644 index 000000000..dea293f40 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,cros-ec-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio codec controlled by ChromeOS EC + +maintainers: + - Cheng-Yi Chiang <cychiang@chromium.org> + - Tzung-Bi Shih <tzungbi@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 inside the "codecs" + subnode of a cros-ec node. + (see Documentation/devicetree/bindings/mfd/google,cros-ec.yaml). + +properties: + compatible: + const: google,cros-ec-codec + + "#sound-dai-cells": + const: 1 + + reg: + items: + - description: | + Physical base address and length of shared memory region from EC. + It contains 3 unsigned 32-bit integer. The first 2 integers + combine to become an unsigned 64-bit physical address. + The last one integer is the length of the shared memory. + + memory-region: + maxItems: 1 + description: | + Shared memory region to EC. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + +required: + - compatible + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + reserved_mem: reserved-mem@52800000 { + compatible = "shared-dma-pool"; + reg = <0x52800000 0x100000>; + no-map; + }; + spi { + #address-cells = <1>; + #size-cells = <0>; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + reg = <0>; + + codecs { + #address-cells = <2>; + #size-cells = <1>; + + cros_ec_codec: ec-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml new file mode 100644 index 000000000..67ccddd44 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7180-trogdor.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7180-Trogdor ASoC sound card driver + +maintainers: + - Rohit kumar <rohitkr@codeaurora.org> + - Cheng-Yi Chiang <cychiang@chromium.org> + +description: + This binding describes the SC7180 sound card which uses LPASS for audio. + +properties: + compatible: + enum: + - google,sc7180-trogdor + - google,sc7180-coachz + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + dmic-gpios: + maxItems: 1 + description: GPIO for switching between DMICs + +patternProperties: + "^dai-link(@[0-9])?$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + required: + - link-name + - cpu + - codec + + additionalProperties: false + +required: + - compatible + - model + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + + - | + sound { + compatible = "google,sc7180-trogdor"; + model = "sc7180-rt5682-max98357a-2mic"; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR"; + + #address-cells = <1>; + #size-cells = <0>; + + dmic-gpios = <&tlmm 86 0>; + + dai-link@0 { + link-name = "MultiMedia0"; + reg = <0>; + cpu { + sound-dai = <&lpass_cpu 0>; + }; + + codec { + sound-dai = <&alc5682 0>; + }; + }; + + dai-link@1 { + link-name = "MultiMedia1"; + reg = <1>; + cpu { + sound-dai = <&lpass_cpu 1>; + }; + + codec { + sound-dai = <&max98357a>; + }; + }; + + dai-link@2 { + link-name = "MultiMedia2"; + reg = <2>; + cpu { + sound-dai = <&lpass_hdmi 0>; + }; + + codec { + sound-dai = <&msm_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml new file mode 100644 index 000000000..869b40363 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -0,0 +1,180 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7280-herobrine.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7280-Herobrine ASoC sound card driver + +maintainers: + - Srinivasa Rao Mandadapu <srivasam@codeaurora.org> + - Judy Hsiao <judyhsiao@chromium.org> + +description: + This binding describes the SC7280 sound card which uses LPASS for audio. + +properties: + compatible: + enum: + - google,sc7280-herobrine + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: true + + required: + - sound-dai + + additionalProperties: false + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: true + + required: + - sound-dai + + additionalProperties: false + + required: + - link-name + - cpu + - codec + - reg + + additionalProperties: false + +required: + - compatible + - model + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound { + compatible = "google,sc7280-herobrine"; + model = "sc7280-wcd938x-max98360a-4dmic"; + + audio-routing = + "IN1_HPHL", "HPHL_OUT", + "IN2_HPHR", "HPHR_OUT", + "AMIC1", "MIC BIAS1", + "AMIC2", "MIC BIAS2", + "VA DMIC0", "MIC BIAS3", + "VA DMIC1", "MIC BIAS3", + "VA DMIC2", "MIC BIAS4", + "VA DMIC3", "MIC BIAS4", + "TX SWR_ADC0", "ADC1_OUTPUT", + "TX SWR_ADC1", "ADC2_OUTPUT", + "TX SWR_ADC2", "ADC3_OUTPUT", + "TX SWR_DMIC0", "DMIC1_OUTPUT", + "TX SWR_DMIC1", "DMIC2_OUTPUT", + "TX SWR_DMIC2", "DMIC3_OUTPUT", + "TX SWR_DMIC3", "DMIC4_OUTPUT"; + + #address-cells = <1>; + #size-cells = <0>; + + dai-link@0 { + link-name = "WCD Playback"; + reg = <LPASS_CDC_DMA_RX0>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_RX0>; + }; + + codec { + sound-dai = <&wcd938x 0>, <&swr0 0>, <&rxmacro 0>; + }; + }; + dai-link@1 { + link-name = "WCD Capture"; + reg = <LPASS_CDC_DMA_TX3>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_TX3>; + }; + + codec { + sound-dai = <&wcd938x 1>, <&swr1 0>, <&txmacro 0>; + }; + }; + + dai-link@2 { + link-name = "MI2S Playback"; + reg = <MI2S_SECONDARY>; + cpu { + sound-dai = <&lpass_cpu MI2S_SECONDARY>; + }; + + codec { + sound-dai = <&max98360a>; + }; + }; + + dai-link@3 { + link-name = "DMIC Capture"; + reg = <LPASS_CDC_DMA_VA_TX0>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_VA_TX0>; + }; + + codec { + sound-dai = <&vamacro 0>; + }; + }; + + dai-link@5 { + link-name = "DP Playback"; + reg = <LPASS_DP_RX>; + cpu { + sound-dai = <&lpass_cpu LPASS_DP_RX>; + }; + + codec { + sound-dai = <&mdss_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt new file mode 100644 index 000000000..efa32a486 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.txt @@ -0,0 +1,19 @@ +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/hisilicon,hi6210-i2s.txt b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt new file mode 100644 index 000000000..7a296784e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt @@ -0,0 +1,42 @@ +* Hisilicon 6210 i2s controller + +Required properties: + +- compatible: should be one of the following: + - "hisilicon,hi6210-i2s" +- reg: physical base address of the i2s controller unit and length of + memory mapped region. +- interrupts: should contain the i2s interrupt. +- clocks: a list of phandle + clock-specifier pairs, one for each entry + in clock-names. +- clock-names: should contain following: + - "dacodec" + - "i2s-base" +- dmas: DMA specifiers for tx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should be "tx" and "rx" +- hisilicon,sysctrl-syscon: phandle to sysctrl syscon +- #sound-dai-cells: Should be set to 1 (for multi-dai) + - The dai cell indexes reference the following interfaces: + 0: S2 interface + (Currently that is the only one available, but more may be + supported in the future) + +Example for the hi6210 i2s controller: + +i2s0: i2s@f7118000{ + compatible = "hisilicon,hi6210-i2s"; + reg = <0x0 0xf7118000 0x0 0x8000>; /* i2s unit */ + interrupts = <GIC_SPI 123 IRQ_TYPE_LEVEL_HIGH>; /* 155 "DigACodec_intr"-32 */ + clocks = <&sys_ctrl HI6220_DACODEC_PCLK>, + <&sys_ctrl HI6220_BBPPLL0_DIV>; + clock-names = "dacodec", "i2s-base"; + dmas = <&dma0 15 &dma0 14>; + dma-names = "rx", "tx"; + hisilicon,sysctrl-syscon = <&sys_ctrl>; + #sound-dai-cells = <1>; +}; + +Then when referencing the i2s controller: + sound-dai = <&i2s0 0>; /* index 0 => S2 interface */ + diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 000000000..e6f05f2f6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,19 @@ +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 +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +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: 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: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/sound/img,i2s-in.txt b/Documentation/devicetree/bindings/sound/img,i2s-in.txt new file mode 100644 index 000000000..423265cfc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,i2s-in.txt @@ -0,0 +1,47 @@ +Imagination Technologies I2S Input Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,i2s-in" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Must include the following entry: + "sys" The system clock + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "rx" Single DMA channel used by all active I2S channels + + - img,i2s-channels : Number of I2S channels instantiated in the I2S in block + +Optional Properties: + + - interrupts : Contains the I2S in interrupts. Depending on + the configuration, there may be no interrupts, one interrupt, + or an interrupt per I2S channel. For the case where there is + one interrupt per channel, the interrupts should be listed + in ascending channel order + + - resets: Contains a phandle to the I2S in reset signal + + - reset-names: Contains the reset signal name "rst" + +Example: + +i2s_in: i2s-in@18100800 { + compatible = "img,i2s-in"; + reg = <0x18100800 0x200>; + interrupts = <GIC_SHARED 7 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 30 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_I2S_IN>; + clock-names = "sys"; + img,i2s-channels = <6>; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,i2s-out.txt b/Documentation/devicetree/bindings/sound/img,i2s-out.txt new file mode 100644 index 000000000..6b0ee9b7e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,i2s-out.txt @@ -0,0 +1,51 @@ +Imagination Technologies I2S Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,i2s-out" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Must include the following entries: + "sys" The system clock + "ref" The reference clock + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" Single DMA channel used by all active I2S channels + + - img,i2s-channels : Number of I2S channels instantiated in the I2S out block + + - resets: Contains a phandle to the I2S out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the I2S out interrupts. Depending on + the configuration, there may be no interrupts, one interrupt, + or an interrupt per I2S channel. For the case where there is + one interrupt per channel, the interrupts should be listed + in ascending channel order + +Example: + +i2s_out: i2s-out@18100a00 { + compatible = "img,i2s-out"; + reg = <0x18100A00 0x200>; + interrupts = <GIC_SHARED 13 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 23 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_I2S_OUT>, + <&clk_core CLK_I2S>; + clock-names = "sys", "ref"; + img,i2s-channels = <6>; + resets = <&pistachio_reset PISTACHIO_RESET_I2S_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,parallel-out.txt b/Documentation/devicetree/bindings/sound/img,parallel-out.txt new file mode 100644 index 000000000..37a3f94cc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,parallel-out.txt @@ -0,0 +1,44 @@ +Imagination Technologies Parallel Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,parallel-out". + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device. + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "sys" The system clock + "ref" The reference clock + + - resets: Contains a phandle to the parallel out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the parallel out interrupt, if present + +Example: + +parallel_out: parallel-out@18100c00 { + compatible = "img,parallel-out"; + reg = <0x18100C00 0x100>; + interrupts = <GIC_SHARED 19 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 16 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_PAUD_OUT>, + <&clk_core CLK_AUDIO_DAC>; + clock-names = "sys", "ref"; + resets = <&pistachio_reset PISTACHIO_RESET_PRL_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt b/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt new file mode 100644 index 000000000..4cc18fc04 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt @@ -0,0 +1,18 @@ +Pistachio internal DAC DT bindings + +Required properties: + + - compatible: "img,pistachio-internal-dac" + + - img,cr-top : Must contain a phandle to the top level control syscon + node which contains the internal dac control registers + + - VDD-supply : Digital power supply regulator (+1.8V or +3.3V) + +Examples: + +internal_dac: internal-dac { + compatible = "img,pistachio-internal-dac"; + img,cr-top = <&cr_top>; + VDD-supply = <&supply3v3>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.txt b/Documentation/devicetree/bindings/sound/img,spdif-in.txt new file mode 100644 index 000000000..f7ea8c87b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,spdif-in.txt @@ -0,0 +1,41 @@ +Imagination Technologies SPDIF Input Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,spdif-in" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "rx" + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Includes the following entries: + "sys" The system clock + +Optional Properties: + + - resets: Should contain a phandle to the spdif in reset signal, if any + + - reset-names: Should contain the reset signal name "rst", if a + reset phandle is given + + - interrupts : Contains the spdif in interrupt, if present + +Example: + +spdif_in: spdif-in@18100e00 { + compatible = "img,spdif-in"; + reg = <0x18100E00 0x100>; + interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 15 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_SPDIF_IN>; + clock-names = "sys"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,spdif-out.txt b/Documentation/devicetree/bindings/sound/img,spdif-out.txt new file mode 100644 index 000000000..413ed8b01 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,spdif-out.txt @@ -0,0 +1,44 @@ +Imagination Technologies SPDIF Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,spdif-out" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "sys" The system clock + "ref" The reference clock + + - resets: Contains a phandle to the spdif out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the parallel out interrupt, if present + +Example: + +spdif_out: spdif-out@18100d00 { + compatible = "img,spdif-out"; + reg = <0x18100D00 0x100>; + interrupts = <GIC_SHARED 21 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 14 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_SPDIF_OUT>, + <&clk_core CLK_SPDIF>; + clock-names = "sys", "ref"; + resets = <&pistachio_reset PISTACHIO_RESET_SPDIF_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; 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 000000000..b6f5d4866 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio sound card. + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + format: + description: audio format. + items: + enum: + - i2s + - dsp_b + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + 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 + - model + +additionalProperties: false + +examples: + - | + sound-ak4458 { + compatible = "fsl,imx-audio-card"; + model = "ak4458-audio"; + pri-dai-link { + link-name = "akcodec"; + format = "i2s"; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + fe-dai-link { + link-name = "HiFi-ASRC-FE"; + format = "i2s"; + cpu { + sound-dai = <&easrc>; + }; + }; + be-dai-link { + link-name = "HiFi-ASRC-BE"; + format = "dsp_b"; + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt new file mode 100644 index 000000000..07b68ab20 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt @@ -0,0 +1,60 @@ +Freescale i.MX audio complex with ES8328 codec + +Required properties: +- compatible : "fsl,imx-audio-es8328" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- jack-gpio : Optional GPIO for headphone jack +- audio-amp-supply : Power regulator for speaker amps +- audio-codec : The phandle of the ES8328 audio codec +- 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, ES8328 + pins, and the jacks on the board: + + Power supplies: + * audio-amp + + ES8328 pins: + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * LINPUT1 + * LINPUT2 + * RINPUT1 + * RINPUT2 + * Mic PGA + + Board connectors: + * Headphone + * Speaker + * Mic Jack +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer (AUDMIX) + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx-audio-es8328"; + model = "imx-audio-es8328"; + ssi-controller = <&ssi1>; + audio-codec = <&codec>; + jack-gpio = <&gpio5 15 0>; + audio-amp-supply = <®_audio_amp>; + audio-routing = + "Speaker", "LOUT2", + "Speaker", "ROUT2", + "Speaker", "audio-amp", + "Headphone", "ROUT1", + "Headphone", "LOUT1", + "LINPUT1", "Mic Jack", + "RINPUT1", "Mic Jack", + "Mic Jack", "Mic Bias"; + mux-int-port = <1>; + mux-ext-port = <3>; +}; 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 000000000..e7e7bb65c --- /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-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt new file mode 100644 index 000000000..2f89db88f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt @@ -0,0 +1,56 @@ +Freescale i.MX audio complex with SGTL5000 codec + +Required properties: + + - compatible : "fsl,imx-audio-sgtl5000" + + - model : The user-visible name of this sound complex + + - ssi-controller : The phandle of the i.MX SSI controller + + - audio-codec : The phandle of the SGTL5000 audio codec + + - 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, SGTL5000 + pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + SGTL5000 pins: + * MIC_IN + * LINE_IN + * HP_OUT + * LINE_OUT + + Board connectors: + * Mic Jack + * Line In Jack + * Headphone Jack + * Line Out Jack + * Ext Spk + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx51-babbage-sgtl5000", + "fsl,imx-audio-sgtl5000"; + model = "imx51-babbage-sgtl5000"; + ssi-controller = <&ssi1>; + audio-codec = <&sgtl5000>; + audio-routing = + "MIC_IN", "Mic Jack", + "Mic Jack", "Mic Bias", + "Headphone Jack", "HP_OUT"; + mux-int-port = <1>; + mux-ext-port = <3>; +}; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 000000000..da84a442c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,36 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the + transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector + or jack on the board or it's connecting to some + other IP block, such as an HDMI encoder or + display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there + is a physical S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000..dab45c310 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel@pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,aic.yaml b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml new file mode 100644 index 000000000..d607325f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml @@ -0,0 +1,92 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ingenic,aic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Ingenic SoCs AC97 / I2S Controller (AIC) DT bindings + +maintainers: + - Paul Cercueil <paul@crapouillou.net> + +properties: + $nodename: + pattern: '^audio-controller@' + + compatible: + oneOf: + - enum: + - ingenic,jz4740-i2s + - ingenic,jz4760-i2s + - ingenic,jz4770-i2s + - ingenic,jz4780-i2s + - items: + - const: ingenic,jz4725b-i2s + - const: ingenic,jz4740-i2s + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: AIC clock + - description: I2S clock + - description: EXT clock + - description: PLL/2 clock + + clock-names: + items: + - const: aic + - const: i2s + - const: ext + - const: pll half + + dmas: + items: + - description: DMA controller phandle and request line for I2S RX + - description: DMA controller phandle and request line for I2S TX + + dma-names: + items: + - const: rx + - const: tx + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> + aic: audio-controller@10020000 { + compatible = "ingenic,jz4740-i2s"; + reg = <0x10020000 0x38>; + + #sound-dai-cells = <0>; + + interrupt-parent = <&intc>; + interrupts = <18>; + + clocks = <&cgu JZ4740_CLK_AIC>, + <&cgu JZ4740_CLK_I2S>, + <&cgu JZ4740_CLK_EXT>, + <&cgu JZ4740_CLK_PLL_HALF>; + clock-names = "aic", "i2s", "ext", "pll half"; + + dmas = <&dmac 25 0xffffffff>, <&dmac 24 0xffffffff>; + dma-names = "rx", "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml new file mode 100644 index 000000000..48aae54dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ingenic,codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Ingenic JZ47xx internal codec DT bindings + +maintainers: + - Paul Cercueil <paul@crapouillou.net> + +properties: + $nodename: + pattern: '^audio-codec@.*' + + compatible: + oneOf: + - enum: + - ingenic,jz4770-codec + - ingenic,jz4760-codec + - ingenic,jz4725b-codec + - ingenic,jz4740-codec + - items: + - const: ingenic,jz4760b-codec + - const: ingenic,jz4760-codec + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: aic + + '#sound-dai-cells': + const: 0 + +additionalProperties: false + +required: + - compatible + - reg + - clocks + - clock-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> + codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + #sound-dai-cells = <0>; + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/inno-rk3036.txt b/Documentation/devicetree/bindings/sound/inno-rk3036.txt new file mode 100644 index 000000000..758de8e27 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/inno-rk3036.txt @@ -0,0 +1,20 @@ +Inno audio codec for RK3036 + +Inno audio codec is integrated inside RK3036 SoC. + +Required properties: +- compatible : Should be "rockchip,rk3036-codec". +- reg : The registers of codec. +- clock-names : Should be "acodec_pclk". +- clocks : The clock of codec. +- rockchip,grf : The phandle of grf device node. + +Example: + + acodec: acodec-ana@20030000 { + compatible = "rk3036-codec"; + reg = <0x20030000 0x4000>; + rockchip,grf = <&grf>; + clock-names = "acodec_pclk"; + clocks = <&cru ACLK_VCODEC>; + }; diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml new file mode 100644 index 000000000..b2603f611 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright 2020 Intel Corporation +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Intel KeemBay I2S + +maintainers: + - Sia, Jee Heng <jee.heng.sia@intel.com> + +description: | + Intel KeemBay I2S + +properties: + compatible: + enum: + - intel,keembay-i2s + - intel,keembay-tdm + - intel,keembay-hdmi-i2s + + "#sound-dai-cells": + const: 0 + + reg: + items: + - description: I2S registers + - description: I2S gen configuration + + reg-names: + items: + - const: i2s-regs + - const: i2s_gen_cfg + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: osc + - const: apb_clk + + dmas: + items: + - description: DMA TX channel + - description: DMA RX channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #define KEEM_BAY_PSS_AUX_I2S3 + #define KEEM_BAY_PSS_I2S3 + i2s3: i2s@20140000 { + compatible = "intel,keembay-i2s"; + #sound-dai-cells = <0>; + reg = <0x20140000 0x200>, /* I2S registers */ + <0x202a00a4 0x4>; /* I2S gen configuration */ + reg-names = "i2s-regs", "i2s_gen_cfg"; + interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; + clock-names = "osc", "apb_clk"; + clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>; + dmas = <&axi_dma0 29>, <&axi_dma0 33>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml new file mode 100644 index 000000000..b97e0fcbd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bluetooth SCO Audio Codec + +maintainers: + - Mark Brown <broonie@kernel.org> + +properties: + '#sound-dai-cells': + enum: + - 0 + + # For Wideband PCM + - 1 + + compatible: + enum: + - delta,dfbmcs320 + - linux,bt-sco + +required: + - '#sound-dai-cells' + - compatible + +additionalProperties: false + +examples: + - | + codec { + #sound-dai-cells = <0>; + compatible = "linux,bt-sco"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml new file mode 100644 index 000000000..808f6d273 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: linux,spdif-dit + + "#sound-dai-cells": + const: 0 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + +additionalProperties: false + +examples: + - | + spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml new file mode 100644 index 000000000..81f266d66 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/marvell,mmp-sspa.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Marvel SSPA Digital Audio Interface Bindings + +maintainers: + - Lubomir Rintel <lkundrak@v3.sk> + +properties: + $nodename: + pattern: "^audio-controller(@.*)?$" + + compatible: + const: marvell,mmp-sspa + + reg: + items: + - description: RX block + - description: TX block + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Clock for the Audio block + - description: I2S bit clock + + clock-names: + items: + - const: audio + - const: bitclk + + power-domains: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + properties: + endpoint: + type: object + + properties: + dai-format: + const: i2s + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - port + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/marvell,mmp2.h> + + audio-controller@d42a0c00 { + compatible = "marvell,mmp-sspa"; + reg = <0xd42a0c00 0x30>, + <0xd42a0c80 0x30>; + interrupts = <2>; + clock-names = "audio", "bitclk"; + clocks = <&soc_clocks 127>, + <&audio_clk 1>; + #sound-dai-cells = <0>; + dmas = <&adma0 0>, <&adma0 1>; + dma-names = "tx", "rx"; + port { + endpoint { + remote-endpoint = <&rt5631_0>; + dai-format = "i2s"; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt new file mode 100644 index 000000000..2ea85d5be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt @@ -0,0 +1,27 @@ +Marvell PXA2xx audio complex + +This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series. + +Required properties: + - compatible: should be one of the following: + "marvell,pxa250-ac97" + "marvell,pxa270-ac97" + "marvell,pxa300-ac97" + - reg: device MMIO address space + - interrupts: single interrupt generated by AC97 IP + - clocks: input clock of the AC97 IP, refer to clock-bindings.txt + +Optional properties: + - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt + - reset-gpios: gpio used for AC97 reset, refer to gpio.txt + +Example: + ac97: sound@40500000 { + compatible = "marvell,pxa250-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pmux_ac97_default >; + }; diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt new file mode 100644 index 000000000..39d640294 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -0,0 +1,59 @@ +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) IRQ_TYPE_LEVEL_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt new file mode 100644 index 000000000..318a4c82f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98095.txt @@ -0,0 +1,22 @@ +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 new file mode 100644 index 000000000..75db84d06 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -0,0 +1,28 @@ +Maxim MAX98357A/MAX98360A audio DAC + +This node models the Maxim MAX98357A/MAX98360A DAC. + +Required properties: +- compatible : "maxim,max98357a" for MAX98357A. + "maxim,max98360a" for MAX98360A. + +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>; +}; + +max98360a { + compatible = "maxim,max98360a"; + sdmode-gpios = <&qcom_pinmux 25 0>; +}; diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 000000000..8b2b2704b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +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/max98373.txt b/Documentation/devicetree/bindings/sound/max98373.txt new file mode 100644 index 000000000..456cb1c59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98373.txt @@ -0,0 +1,40 @@ +Maxim Integrated MAX98373 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98373" + + - reg : the I2C address of the device. + +Optional properties: + + - maxim,vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + slot range : 0 ~ 15, Default : 0 + + - maxim,imon-slot-no : slot number used to send current information + slot range : 0 ~ 15, Default : 0 + + - maxim,spkfb-slot-no : slot number used to send speaker feedback information + slot range : 0 ~ 15, Default : 0 + + - maxim,interleave-mode : 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. + Boolean, define to enable the interleave mode, Default : false + +Example: + +codec: max98373@31 { + compatible = "maxim,max98373"; + reg = <0x31>; + maxim,vmon-slot-no = <0>; + maxim,imon-slot-no = <1>; + maxim,spkfb-slot-no = <2>; + maxim,interleave-mode; +}; diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt new file mode 100644 index 000000000..583ed5fdf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98504.txt @@ -0,0 +1,44 @@ +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/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt new file mode 100644 index 000000000..e0d4e95e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9860.txt @@ -0,0 +1,28 @@ +MAX9860 Mono Audio Voice Codec + +Required properties: + + - compatible : "maxim,max9860" + + - reg : the I2C address of the device + + - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for + the device, as covered in bindings/regulator/regulator.txt + + - clock-names : Required element: "mclk". + + - clocks : A clock specifier for the clock connected as MCLK. + +Examples: + + max9860: max9860@10 { + compatible = "maxim,max9860"; + reg = <0x10>; + + AVDD-supply = <®_1v8>; + DVDD-supply = <®_1v8>; + DVDDIO-supply = <®_3v0>; + + clock-names = "mclk"; + clocks = <&pck2>; + }; diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt new file mode 100644 index 000000000..b8bd914ee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9867.txt @@ -0,0 +1,17 @@ +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 new file mode 100644 index 000000000..98cb9ba5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -0,0 +1,44 @@ +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 + +Optional properties: + - reset-gpios : GPIO to reset the device + +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 new file mode 100644 index 000000000..737a99637 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9759.txt @@ -0,0 +1,18 @@ +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,max98088.txt b/Documentation/devicetree/bindings/sound/maxim,max98088.txt new file mode 100644 index 000000000..da764d913 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98088.txt @@ -0,0 +1,23 @@ +MAX98088 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible: "maxim,max98088" or "maxim,max98089". +- reg: The I2C address of the device. + +Optional properties: + +- clocks: the clock provider of MCLK, see ../clock/clock-bindings.txt section + "consumer" for more information. +- clock-names: must be set to "mclk" + +Example: + +max98089: codec@10 { + compatible = "maxim,max98089"; + reg = <0x10>; + clocks = <&clks IMX6QDL_CLK_CKO2>; + clock-names = "mclk"; +}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000..deaa6886c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management + +maintainers: + - Steve Lee <steves.lee@maximintegrated.com> + +properties: + compatible: + const: maxim,max98390 + + reg: + maxItems: 1 + description: I2C address of the device. + + maxim,temperature_calib: + description: The calculated temperature data was measured while doing the calibration. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 65535 + + maxim,r0_calib: + description: This is r0 calibration data which was measured in factory mode. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 8388607 + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + reset-gpios = <&gpio 9 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000..3f88c7d61 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song@maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml b/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml new file mode 100644 index 000000000..0481315cb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,i2s-mcc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip I2S Multi-Channel Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and + supports a Time Division Multiplexed (TDM) interface with external + multi-channel audio codecs. It consists of a receiver, a transmitter and a + common clock generator that can be enabled separately to provide Adapter, + Client or Controller modes with receiver and/or transmitter active. + On later I2SMCC versions (starting with Microchip's SAMA7G5) I2S + multi-channel is supported by using multiple data pins, output and + input, without TDM. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - microchip,sam9x60-i2smcc + - microchip,sama7g5-i2smcc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock (Optional). Should be set mostly when Master + Mode is required. + minItems: 1 + + clock-names: + items: + - const: pclk + - const: gclk + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + microchip,tdm-data-pair: + description: + Represents the DIN/DOUT pair pins that are used to receive/send + TDM data. It is optional and it is only needed if the controller + uses the TDM mode. + $ref: /schemas/types.yaml#/definitions/uint8 + enum: [0, 1, 2, 3] + default: 0 + +if: + properties: + compatible: + const: microchip,sam9x60-i2smcc +then: + properties: + microchip,tdm-data-pair: false + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f001c000 { + #sound-dai-cells = <0>; + compatible = "microchip,sam9x60-i2smcc"; + reg = <0xf001c000 0x100>; + interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(36))>, + <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(37))>; + dma-names = "tx", "rx"; + clocks = <&i2s_clk>, <&i2s_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml new file mode 100644 index 000000000..70a47c682 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Rx Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Receiver is a serial port + compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdifrx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdifrx: spdifrx@e1614000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdifrx"; + reg = <0xe1614000 0x4000>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(49)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 84>, <&pmc PMC_TYPE_GCK 84>; + clock-names = "pclk", "gclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000..d218e4ab9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a serial port + compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + maxItems: 1 + + dma-names: + const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml new file mode 100644 index 000000000..04414eb4a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,pdmc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip Pulse Density Microphone Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Pulse Density Microphone Controller (PDMC) interfaces up to 4 + digital microphones having Pulse Density Modulated (PDM) outputs. + +properties: + compatible: + const: microchip,sama7g5-pdmc + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + + microchip,mic-pos: + description: | + Position of PDM microphones on the DS line and the sampling edge (rising + or falling) of the CLK line. A microphone is represented as a pair of DS + line and the sampling edge. The first microphone is mapped to channel 0, + the second to channel 1, etc. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: value for DS line + - description: value for sampling edge + anyOf: + - enum: + - [0, 0] + - [0, 1] + - [1, 0] + - [1, 1] + minItems: 1 + maxItems: 4 + uniqueItems: true + +required: + - compatible + - reg + - "#sound-dai-cells" + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - microchip,mic-pos + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/sound/microchip,pdmc.h> + + pdmc: sound@e1608000 { + compatible = "microchip,sama7g5-pdmc"; + reg = <0xe1608000 0x4000>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 68 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(37)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 68>, <&pmc PMC_TYPE_GCK 68>; + clock-names = "pclk", "gclk"; + microchip,mic-pos = <MCHP_PDMC_DS0 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS0 MCHP_PDMC_CLK_NEGATIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_NEGATIVE>; + }; diff --git a/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt new file mode 100644 index 000000000..912f8fae1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt @@ -0,0 +1,23 @@ +Mikroe-PROTO audio board + +Required properties: + - compatible: "mikroe,mikroe-proto" + - dai-format: Must be "i2s". + - i2s-controller: The phandle of the I2S controller. + - audio-codec: The phandle of the WM8731 audio codec. +Optional properties: + - model: The user-visible name of this sound complex. + - bitclock-master: Indicates dai-link bit clock master; for details see simple-card.txt (1). + - frame-master: Indicates dai-link frame master; for details see simple-card.txt (1). + +(1) : There must be the same master for both bit and frame clocks. + +Example: + sound { + compatible = "mikroe,mikroe-proto"; + model = "wm8731 @ sama5d2_xplained"; + i2s-controller = <&i2s0>; + audio-codec = <&wm8731>; + dai-format = "i2s"; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt new file mode 100644 index 000000000..feef39b4a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -0,0 +1,34 @@ +Marvell PXA SSP CPU DAI bindings + +Required properties: + + compatible Must be "mrvl,pxa-ssp-dai" + port A phandle reference to a PXA ssp upstream device + +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + +Example: + + /* upstream device */ + + ssp1: ssp@41000000 { + compatible = "mrvl,pxa3xx-ssp"; + reg = <0x41000000 0x40>; + interrupts = <24>; + clock-names = "pxa27x-ssp.0"; + }; + + /* DAI as user */ + + ssp_dai0: ssp_dai@0 { + compatible = "mrvl,pxa-ssp-dai"; + port = <&ssp1>; + #sound-dai-cells = <0>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt new file mode 100644 index 000000000..560762e0a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt @@ -0,0 +1,146 @@ +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/mt2701-cs42448.txt b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt new file mode 100644 index 000000000..05574446c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt @@ -0,0 +1,43 @@ +MT2701 with CS42448 CODEC + +Required properties: +- compatible: "mediatek,mt2701-cs42448-machine" +- mediatek,platform: the phandle of MT2701 ASoC platform +- audio-routing: a list of the connections between audio +- mediatek,audio-codec: the phandles of cs42448 codec +- mediatek,audio-codec-bt-mrg the phandles of bt-sco dummy codec +- pinctrl-names: Should contain only one value - "default" +- pinctrl-0: Should specify pin control groups used for this controller. +- i2s1-in-sel-gpio1, i2s1-in-sel-gpio2: Should specify two gpio pins to + control I2S1-in mux. + +Example: + + sound:sound { + compatible = "mediatek,mt2701-cs42448-machine"; + mediatek,platform = <&afe>; + /* CS42448 Machine name */ + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "AMIC", + "AIN1R", "AMIC", + "AIN2L", "Tuner In", + "AIN2R", "Tuner In", + "AIN3L", "Satellite Tuner In", + "AIN3R", "Satellite Tuner In", + "AIN3L", "AUX In", + "AIN3R", "AUX In"; + mediatek,audio-codec = <&cs42448>; + mediatek,audio-codec-bt-mrg = <&bt_sco_codec>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + i2s1-in-sel-gpio1 = <&pio 53 0>; + i2s1-in-sel-gpio2 = <&pio 54 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt new file mode 100644 index 000000000..809b609ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt @@ -0,0 +1,24 @@ +MT2701 with WM8960 CODEC + +Required properties: +- compatible: "mediatek,mt2701-wm8960-machine" +- mediatek,platform: the phandle of MT2701 ASoC platform +- audio-routing: a list of the connections between audio +- mediatek,audio-codec: the phandles of wm8960 codec +- pinctrl-names: Should contain only one value - "default" +- pinctrl-0: Should specify pin control groups used for this controller. + +Example: + + sound:sound { + compatible = "mediatek,mt2701-wm8960-machine"; + mediatek,platform = <&afe>; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + mediatek,audio-codec = <&wm8960>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt6351.txt b/Documentation/devicetree/bindings/sound/mt6351.txt new file mode 100644 index 000000000..7fb2cb992 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6351.txt @@ -0,0 +1,16 @@ +Mediatek MT6351 Audio Codec + +The communication between MT6351 and SoC is through Mediatek PMIC wrapper. +For more detail, please visit Mediatek PMIC wrapper documentation. + +Must be a child node of PMIC wrapper. + +Required properties: + +- compatible : "mediatek,mt6351-sound". + +Example: + +mt6351_snd { + compatible = "mediatek,mt6351-sound"; +}; diff --git a/Documentation/devicetree/bindings/sound/mt6358.txt b/Documentation/devicetree/bindings/sound/mt6358.txt new file mode 100644 index 000000000..fbe9e55c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6358.txt @@ -0,0 +1,26 @@ +Mediatek MT6358 Audio Codec + +The communication between MT6358 and SoC is through Mediatek PMIC wrapper. +For more detail, please visit Mediatek PMIC wrapper documentation. + +Must be a child node of PMIC wrapper. + +Required properties: + +- compatible - "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 000000000..23d411fc4 --- /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/mt6797-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt new file mode 100644 index 000000000..0ae29de15 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt @@ -0,0 +1,42 @@ +Mediatek AFE PCM controller for mt6797 + +Required properties: +- compatible = "mediatek,mt6797-audio"; +- reg: register location and size +- interrupts: should contain AFE interrupt +- power-domains: should define the power domain +- clocks: Must contain an entry for each entry in clock-names +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "infra_sys_audio_26m", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_sys_pll1_d4", + "top_clk26m_clk"; + +Example: + + afe: mt6797-afe-pcm@11220000 { + compatible = "mediatek,mt6797-audio"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 151 IRQ_TYPE_LEVEL_LOW>; + power-domains = <&scpsys MT6797_POWER_DOMAIN_AUDIO>; + clocks = <&infrasys CLK_INFRA_AUDIO>, + <&infrasys CLK_INFRA_AUDIO_26M>, + <&infrasys CLK_INFRA_AUDIO_26M_PAD_TOP>, + <&topckgen CLK_TOP_MUX_AUDIO>, + <&topckgen CLK_TOP_MUX_AUD_INTBUS>, + <&topckgen CLK_TOP_SYSPLL3_D4>, + <&topckgen CLK_TOP_SYSPLL1_D4>, + <&clk26m>; + clock-names = "infra_sys_audio_clk", + "infra_sys_audio_26m", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_sys_pll1_d4", + "top_clk26m_clk"; + }; diff --git a/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt b/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt new file mode 100644 index 000000000..1d95a8840 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt @@ -0,0 +1,14 @@ +MT6797 with MT6351 CODEC + +Required properties: +- compatible: "mediatek,mt6797-mt6351-sound" +- mediatek,platform: the phandle of MT6797 ASoC platform +- mediatek,audio-codec: the phandles of MT6351 codec + +Example: + + sound { + compatible = "mediatek,mt6797-mt6351-sound"; + mediatek,audio-codec = <&mt6351_snd>; + mediatek,platform = <&afe>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt new file mode 100644 index 000000000..519e97c8f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -0,0 +1,15 @@ +MT8173 with MAX98090 CODEC + +Required properties: +- compatible : "mediatek,mt8173-max98090" +- mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-max98090"; + mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt new file mode 100644 index 000000000..e8b3c80c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 RT5514 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5514" +- mediatek,audio-codec: the phandles of rt5650 and rt5514 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5514"; + mediatek,audio-codec = <&rt5650 &rt5514>; + mediatek,platform = <&afe>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt new file mode 100644 index 000000000..ac28cdb49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -0,0 +1,16 @@ +MT8173 with RT5650 RT5676 CODECS and HDMI via I2S + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5676" +- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs + and of the hdmi encoder node +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5676"; + mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>; + mediatek,platform = <&afe>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt new file mode 100644 index 000000000..29dce2ac8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -0,0 +1,31 @@ +MT8173 with RT5650 CODECS and HDMI via I2S + +Required properties: +- compatible : "mediatek,mt8173-rt5650" +- mediatek,audio-codec: the phandles of rt5650 codecs + and of the hdmi encoder node +- mediatek,platform: the phandle of MT8173 ASoC platform + +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + +- mediatek,mclk: the MCLK source + 0 : external oscillator, MCLK = 12.288M + 1 : internal source from mt8173, MCLK = sampling rate*256 + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650"; + mediatek,audio-codec = <&rt5650 &hdmi0>; + mediatek,platform = <&afe>; + mediatek,mclk = <0>; + codec-capture { + sound-dai = <&rt5650 1>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt new file mode 100644 index 000000000..1f1cba415 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt @@ -0,0 +1,42 @@ +Mediatek AFE PCM controller for mt8183 + +Required properties: +- compatible = "mediatek,mt68183-audio"; +- reg: register location and size +- interrupts: should contain AFE interrupt +- resets: Must contain an entry for each entry in reset-names + See ../reset/reset.txt for details. +- reset-names: should have these reset names: + "audiosys"; +- power-domains: should define the power domain +- clocks: Must contain an entry for each entry in clock-names +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_clk26m_clk"; + +Example: + + afe: mt8183-afe-pcm@11220000 { + compatible = "mediatek,mt8183-audio"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>; + resets = <&watchdog MT8183_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + power-domains = <&scpsys MT8183_POWER_DOMAIN_AUDIO>; + clocks = <&infrasys CLK_INFRA_AUDIO>, + <&infrasys CLK_INFRA_AUDIO_26M_BCLK>, + <&topckgen CLK_TOP_MUX_AUDIO>, + <&topckgen CLK_TOP_MUX_AUD_INTBUS>, + <&topckgen CLK_TOP_SYSPLL_D2_D4>, + <&clk26m>; + clock-names = "infra_sys_audio_clk", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll_d2_d4", + "top_clk26m_clk"; + }; diff --git a/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt new file mode 100644 index 000000000..f276dfc74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt @@ -0,0 +1,21 @@ +MT8183 with MT6358, DA7219, MAX98357, and RT1015 CODECS + +Required properties: +- 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 new file mode 100644 index 000000000..ecd46ed8e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -0,0 +1,25 @@ +MT8183 with MT6358, TS3A227, MAX98357, and RT1015 CODECS + +Required properties: +- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" for MAX98357A codec + "mediatek,mt8183_mt6358_ts3a227_max98357b" for MAX98357B codec + "mediatek,mt8183_mt6358_ts3a227_rt1015" for RT1015 codec + "mediatek,mt8183_mt6358_ts3a227_rt1015p" for RT1015P codec +- mediatek,platform: the phandle of MT8183 ASoC platform + +Optional properties: +- mediatek,headset-codec: the phandles of ts3a227 codecs +- mediatek,ec-codec: the phandle of EC codecs. + See google,cros-ec-codec.txt for more details. +- mediatek,hdmi-codec: the phandles of HDMI codec + +Example: + + sound { + compatible = "mediatek,mt8183_mt6358_ts3a227_max98357"; + mediatek,headset-codec = <&ts3a227>; + mediatek,ec-codec = <&ec_codec>; + mediatek,hdmi-codec = <&it6505dptx>; + mediatek,platform = <&afe>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml new file mode 100644 index 000000000..88f82d096 --- /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 000000000..d427f7f62 --- /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: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - playback-codecs + +examples: + - | + + sound: mt8186-sound { + compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound"; + mediatek,platform = <&afe>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + + headset-codec { + sound-dai = <&da7219>; + }; + + playback-codecs { + sound-dai = <&anx_bridge_dp>, + <&max98357a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml new file mode 100644 index 000000000..4fc5b045d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8186-mt6366-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,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: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - playback-codecs + +examples: + - | + + sound: mt8186-sound { + compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound"; + mediatek,platform = <&afe>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + + headset-codec { + sound-dai = <&rt5682s>; + }; + + playback-codecs { + sound-dai = <&it6505dptx>, + <&rt1019p>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000..7a25bc9b8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 000000000..478be7e3f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.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/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: + $ref: /schemas/types.yaml#/definitions/phandle + required: + - sound-dai + + speaker-codecs: + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 2 + items: + maxItems: 1 + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - sound-dai + +additionalProperties: false + +required: + - compatible + - mediatek,platform + - headset-codec + - speaker-codecs + +examples: + - | + + sound: mt8192-sound { + compatible = "mediatek,mt8192_mt6359_rt1015_rt5682"; + mediatek,platform = <&afe>; + mediatek,hdmi-codec = <&anx_bridge_dp>; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + + headset-codec { + sound-dai = <&rt5682>; + }; + + speaker-codecs { + sound-dai = <&rt1015_l>, + <&rt1015_r>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml new file mode 100644 index 000000000..4452a4070 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml @@ -0,0 +1,200 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8195-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8195 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8195-audio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + memory-region: + maxItems: 1 + description: | + Shared memory region for AFE memif. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: 26M clock + - description: audio pll1 clock + - description: audio pll2 clock + - description: clock divider for i2si1_mck + - description: clock divider for i2si2_mck + - description: clock divider for i2so1_mck + - description: clock divider for i2so2_mck + - description: clock divider for dptx_mck + - description: a1sys hoping clock + - description: audio intbus clock + - description: audio hires clock + - description: audio local bus clock + - description: mux for dptx_mck + - description: mux for i2so1_mck + - description: mux for i2so2_mck + - description: mux for i2si1_mck + - description: mux for i2si2_mck + - description: audio infra 26M clock + - description: infra bus clock + + clock-names: + items: + - const: clk26m + - const: apll1_ck + - const: apll2_ck + - const: apll12_div0 + - const: apll12_div1 + - const: apll12_div2 + - const: apll12_div3 + - const: apll12_div9 + - const: a1sys_hp_sel + - const: aud_intbus_sel + - const: audio_h_sel + - const: audio_local_bus_sel + - const: dptx_m_sel + - const: i2so1_m_sel + - const: i2so2_m_sel + - const: i2si1_m_sel + - const: i2si2_m_sel + - const: infra_ao_audio_26m_b + - const: scp_adsp_audiodsp + + mediatek,etdm-in1-chn-disabled: + $ref: /schemas/types.yaml#/definitions/uint8-array + maxItems: 24 + description: Specify which input channel should be disabled. + + mediatek,etdm-in2-chn-disabled: + $ref: /schemas/types.yaml#/definitions/uint8-array + maxItems: 16 + description: Specify which input channel should be disabled. + +patternProperties: + "^mediatek,etdm-in[1-2]-mclk-always-on-rate-hz$": + description: Specify etdm in mclk output rate for always on case. + + "^mediatek,etdm-out[1-3]-mclk-always-on-rate-hz$": + description: Specify etdm out mclk output rate for always on case. + + "^mediatek,etdm-in[1-2]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-out[1-3]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-in[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in moudule. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + "^mediatek,etdm-out[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out moudule. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + +required: + - compatible + - reg + - interrupts + - resets + - reset-names + - mediatek,topckgen + - power-domains + - clocks + - clock-names + - memory-region + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe: mt8195-afe-pcm@10890000 { + compatible = "mediatek,mt8195-audio"; + reg = <0x10890000 0x10000>; + interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; + resets = <&watchdog 14>; + reset-names = "audiosys"; + mediatek,topckgen = <&topckgen>; + power-domains = <&spm 7>; //MT8195_POWER_DOMAIN_AUDIO + memory-region = <&snd_dma_mem_reserved>; + clocks = <&clk26m>, + <&topckgen 163>, //CLK_TOP_APLL1 + <&topckgen 166>, //CLK_TOP_APLL2 + <&topckgen 233>, //CLK_TOP_APLL12_DIV0 + <&topckgen 234>, //CLK_TOP_APLL12_DIV1 + <&topckgen 235>, //CLK_TOP_APLL12_DIV2 + <&topckgen 236>, //CLK_TOP_APLL12_DIV3 + <&topckgen 238>, //CLK_TOP_APLL12_DIV9 + <&topckgen 100>, //CLK_TOP_A1SYS_HP_SEL + <&topckgen 33>, //CLK_TOP_AUD_INTBUS_SEL + <&topckgen 34>, //CLK_TOP_AUDIO_H_SEL + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS_SEL + <&topckgen 98>, //CLK_TOP_DPTX_M_SEL + <&topckgen 94>, //CLK_TOP_I2SO1_M_SEL + <&topckgen 95>, //CLK_TOP_I2SO2_M_SEL + <&topckgen 96>, //CLK_TOP_I2SI1_M_SEL + <&topckgen 97>, //CLK_TOP_I2SI2_M_SEL + <&infracfg_ao 50>, //CLK_INFRA_AO_AUDIO_26M_B + <&scp_adsp 0>; //CLK_SCP_ADSP_AUDIODSP + clock-names = "clk26m", + "apll1_ck", + "apll2_ck", + "apll12_div0", + "apll12_div1", + "apll12_div2", + "apll12_div3", + "apll12_div9", + "a1sys_hp_sel", + "aud_intbus_sel", + "audio_h_sel", + "audio_local_bus_sel", + "dptx_m_sel", + "i2so1_m_sel", + "i2so2_m_sel", + "i2si1_m_sel", + "i2si2_m_sel", + "infra_ao_audio_26m_b", + "scp_adsp_audiodsp"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml new file mode 100644 index 000000000..ad3447ff8 --- /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/mtk-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt new file mode 100644 index 000000000..e302c7f43 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt @@ -0,0 +1,45 @@ +Mediatek AFE PCM controller + +Required properties: +- compatible = "mediatek,mt8173-afe-pcm"; +- reg: register location and size +- interrupts: Should contain AFE interrupt +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + +Example: + + afe: mt8173-afe-pcm@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + clocks = <&infracfg INFRA_AUDIO>, + <&topckgen TOP_AUDIO_SEL>, + <&topckgen TOP_AUD_INTBUS_SEL>, + <&topckgen TOP_APLL1_DIV0>, + <&topckgen TOP_APLL2_DIV0>, + <&topckgen TOP_I2S0_M_CK_SEL>, + <&topckgen TOP_I2S1_M_CK_SEL>, + <&topckgen TOP_I2S2_M_CK_SEL>, + <&topckgen TOP_I2S3_M_CK_SEL>, + <&topckgen TOP_I2S3_B_CK_SEL>; + clock-names = "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + }; diff --git a/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt new file mode 100644 index 000000000..679e44839 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt @@ -0,0 +1,24 @@ +Mediatek ALSA BT SCO CVSD/MSBC Driver + +Required properties: +- compatible = "mediatek,mtk-btcvsd-snd"; +- reg: register location and size of PKV and SRAM_BANK2 +- interrupts: should contain BTSCO interrupt +- mediatek,infracfg: the phandles of INFRASYS +- mediatek,offset: Array contains of register offset and mask + infra_misc_offset, + infra_conn_bt_cvsd_mask, + cvsd_mcu_read_offset, + cvsd_mcu_write_offset, + cvsd_packet_indicator_offset + +Example: + + mtk-btcvsd-snd@18000000 { + compatible = "mediatek,mtk-btcvsd-snd"; + reg=<0 0x18000000 0 0x1000>, + <0 0x18080000 0 0x8000>; + interrupts = <GIC_SPI 286 IRQ_TYPE_LEVEL_LOW>; + mediatek,infracfg = <&infrasys>; + mediatek,offset = <0xf00 0x800 0xfd0 0xfd4 0xfd8>; + }; diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 000000000..cb8c07c81 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,34 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: + "marvell,kirkwood-audio" for Kirkwood platforms + "marvell,dove-audio" for Dove platforms + "marvell,armada370-audio" for Armada 370 platforms + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: + with "marvell,kirkwood-audio", the audio interrupt + with "marvell,dove-audio", a list of two interrupts, the first for + the data flow, and the second for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,dove-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt new file mode 100644 index 000000000..4eb980bd0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt @@ -0,0 +1,42 @@ +* Freescale MXS audio complex with SGTL5000 codec + +Required properties: +- compatible : "fsl,mxs-audio-sgtl5000" +- model : The user-visible name of this sound complex +- saif-controllers : The phandle list of the MXS SAIF controller +- audio-codec : The phandle of the SGTL5000 audio codec +- 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, SGTL5000 + pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + SGTL5000 pins: + * MIC_IN + * LINE_IN + * HP_OUT + * LINE_OUT + + Board connectors: + * Mic Jack + * Line In Jack + * Headphone Jack + * Line Out Jack + * Ext Spk + +Example: + +sound { + compatible = "fsl,imx28-evk-sgtl5000", + "fsl,mxs-audio-sgtl5000"; + model = "imx28-evk-sgtl5000"; + saif-controllers = <&saif0 &saif1>; + audio-codec = <&sgtl5000>; + audio-routing = + "MIC_IN", "Mic Jack", + "Mic Jack", "Mic Bias", + "Headphone Jack", "HP_OUT"; +}; diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt new file mode 100644 index 000000000..7ba07a118 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -0,0 +1,41 @@ +* Freescale MXS Serial Audio Interface (SAIF) + +Required properties: +- compatible: Should be "fsl,<chip>-saif" +- reg: Should contain registers location and length +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "rx-tx". + +Optional properties: +- fsl,saif-master: phandle to the master SAIF. It's only required for + the slave SAIF. + +Note: Each SAIF controller should have an alias correctly numbered +in "aliases" node. + +Example: + +aliases { + saif0 = &saif0; + saif1 = &saif1; +}; + +saif0: saif@80042000 { + compatible = "fsl,imx28-saif"; + reg = <0x80042000 2000>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; +}; + +saif1: saif@80046000 { + compatible = "fsl,imx28-saif"; + reg = <0x80046000 2000>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; + fsl,saif-master = <&saif0>; +}; diff --git a/Documentation/devicetree/bindings/sound/name-prefix.yaml b/Documentation/devicetree/bindings/sound/name-prefix.yaml new file mode 100644 index 000000000..2fe57f87a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/name-prefix.yaml @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/name-prefix.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Component sound name prefix + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: | + Card implementing the routing property define the connection between + audio components as list of string pair. Component using the same + sink/source names may use this property to prepend the name of their + sinks/sources with the provided string. + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt new file mode 100644 index 000000000..6eaec46f3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8315.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8315 Mono Class-D Amplifier + +Required properties: +- compatible : "nuvoton,nau8315" + +Optional properties: +- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +#include <dt-bindings/gpio/gpio.h> + +nau8315 { + compatible = "nuvoton,nau8315"; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/nau8540.txt b/Documentation/devicetree/bindings/sound/nau8540.txt new file mode 100644 index 000000000..307a76528 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8540.txt @@ -0,0 +1,16 @@ +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 new file mode 100644 index 000000000..7deaa452b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,17 @@ +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/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt new file mode 100644 index 000000000..7c84e7c73 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -0,0 +1,55 @@ +Nuvoton NAU88L21 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8821" + + - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. + - nuvoton,key_enable: Headset button detection switch. + +Example: + + headset: nau8821@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,dmic-clk-threshold = 3072000; + }; diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt new file mode 100644 index 000000000..a471d162d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8822.txt @@ -0,0 +1,16 @@ +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 new file mode 100644 index 000000000..e0058b97e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8824.txt @@ -0,0 +1,88 @@ +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 new file mode 100644 index 000000000..cb861aca8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,108 @@ +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. + + - nuvoton,adcout-drive-strong: make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + + - 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/nokia,rx51.txt b/Documentation/devicetree/bindings/sound/nokia,rx51.txt new file mode 100644 index 000000000..72f93d996 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nokia,rx51.txt @@ -0,0 +1,27 @@ +* Nokia N900 audio setup + +Required properties: +- compatible: Should contain "nokia,n900-audio" +- nokia,cpu-dai: phandle for the McBSP node +- nokia,audio-codec: phandles for the main TLV320AIC3X node and the + auxiliary TLV320AIC3X node (in this order) +- nokia,headphone-amplifier: phandle for the TPA6130A2 node +- tvout-selection-gpios: GPIO for tvout selection +- jack-detection-gpios: GPIO for jack detection +- eci-switch-gpios: GPIO for ECI (Enhancement Control Interface) switch +- speaker-amplifier-gpios: GPIO for speaker amplifier + +Example: + +sound { + compatible = "nokia,n900-audio"; + + nokia,cpu-dai = <&mcbsp2>; + nokia,audio-codec = <&tlv320aic3x>, <&tlv320aic3x_aux>; + nokia,headphone-amplifier = <&tpa6130a2>; + + tvout-selection-gpios = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */ + jack-detection-gpios = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */ + eci-switch-gpios = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */ + speaker-amplifier-gpios = <&twl_gpio 7 GPIO_ACTIVE_HIGH>; +}; 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 000000000..7ef774910 --- /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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headset Stereophone" + - "Int Spk" + - "Headset Mic" + - "Digital Mic" + + # CODEC Pins + - SPKOUT + - SPKOUTN + - MICBIAS1 + - MIC1 + - HPR + - HPL + - DMICDAT + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "Headset Mic", "MICBIAS1", + "MIC1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL", + "DMICDAT", "Digital Mic"; + + nvidia,i2s-controller = <&i2s>; + nvidia,audio-codec = <&codec>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml new file mode 100644 index 000000000..82801b4f4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/nvidia,tegra-audio-common.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Common properties for NVIDIA Tegra audio complexes + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +properties: + clocks: + items: + - description: PLL A clock + - description: PLL A OUT0 clock + - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk + + clock-names: + items: + - const: pll_a + - const: pll_a_out0 + - const: mclk + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + + nvidia,ac97-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the AC97 controller + + nvidia,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the Tegra I2S controller + + nvidia,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,ear-sel-gpios: + maxItems: 1 + description: The GPIO that switch between the microphones + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml new file mode 100644 index 000000000..b4bee466d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -0,0 +1,199 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-graph-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph based Tegra sound card driver + +description: | + This is based on generic audio graph card driver along with additional + customizations for Tegra platforms. It uses the same bindings with + additional standard clock DT bindings required for Tegra. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +allOf: + - $ref: audio-graph.yaml# + +properties: + compatible: + enum: + - nvidia,tegra210-audio-graph-card + - nvidia,tegra186-audio-graph-card + + clocks: + minItems: 2 + + clock-names: + items: + - const: pll_a + - const: plla_out0 + + assigned-clocks: + minItems: 1 + maxItems: 3 + + assigned-clock-parents: + minItems: 1 + maxItems: 3 + + assigned-clock-rates: + minItems: 1 + maxItems: 3 + + interconnects: + items: + - description: APE read memory client + - description: APE write memory client + + interconnect-names: + items: + - const: dma-mem # read + - const: write + + iommus: + maxItems: 1 + +required: + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +unevaluatedProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + tegra_sound { + compatible = "nvidia,tegra210-audio-graph-card"; + + clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + clock-names = "pll_a", "plla_out0"; + + assigned-clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA210_CLK_EXTERN1>; + assigned-clock-parents = <0>, <0>, <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <368640000>, <49152000>, <12288000>; + + dais = /* FE */ + <&admaif1_port>, + /* Router */ + <&xbar_i2s1_port>, + /* I/O DAP Ports */ + <&i2s1_port>; + + label = "jetson-tx1-ape"; + }; + + // The ports are defined for AHUB and its child devices. + ahub@702d0800 { + compatible = "nvidia,tegra210-ahub"; + reg = <0x702d0800 0x800>; + clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + clock-names = "ahub"; + assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0x702d0000 0x702d0000 0x0000e400>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + xbar_admaif1_ep: endpoint { + remote-endpoint = <&admaif1_ep>; + }; + }; + + // ... + + xbar_i2s1_port: port@a { + reg = <0xa>; + xbar_i2s1_ep: endpoint { + remote-endpoint = <&i2s1_cif_ep>; + }; + }; + }; + + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + admaif1_port: port@0 { + reg = <0x0>; + admaif1_ep: endpoint { + remote-endpoint = <&xbar_admaif1_ep>; + }; + }; + + // More ADMAIF ports to follow + }; + }; + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + reg = <0x702d1000 0x100>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + + i2s1_cif_ep: endpoint { + remote-endpoint = <&xbar_i2s1_ep>; + }; + }; + + i2s1_port: port@1 { + reg = <0x1>; + + i2s1_dap: endpoint { + dai-format = "i2s"; + }; + }; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml new file mode 100644 index 000000000..ccc2ee77c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml @@ -0,0 +1,97 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with MAX98090 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98090(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-max98090 + - items: + - enum: + - nvidia,tegra-audio-max98090-nyan-big + - nvidia,tegra-audio-max98090-nyan-blaze + - const: nvidia,tegra-audio-max98090-nyan + - const: nvidia,tegra-audio-max98090 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphones" + - "Speakers" + - "Mic Jack" + - "Int Mic" + + # CODEC Pins + - MIC1 + - MIC2 + - DMICL + - DMICR + - IN1 + - IN2 + - IN3 + - IN4 + - IN5 + - IN6 + - IN12 + - IN34 + - IN56 + - HPL + - HPR + - SPKL + - SPKR + - RCVL + - RCVR + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra124-car.h> + + sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml new file mode 100644 index 000000000..b1deaf271 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphones" + - "Speakers" + - "Mic Jack" + + # CODEC Pins + - DMIC1 + - DMIC2 + - MICBIAS1 + - IN1P + - IN1R + - IN2P + - IN2R + - HPOL + - HPOR + - LOUTL + - LOUTR + - MONOP + - MONON + - SPOLP + - SPOLN + - SPORP + - SPORN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5640-dalmore", + "nvidia,tegra-audio-rt5640"; + nvidia,model = "NVIDIA Tegra Dalmore"; + + nvidia,audio-routing = + "Headphones", "HPOR", + "Headphones", "HPOL", + "Speakers", "SPORP", + "Speakers", "SPORN", + "Speakers", "SPOLP", + "Speakers", "SPOLN"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5640>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, <&clk 217>, <&clk 120>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml new file mode 100644 index 000000000..a49997d60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone" + - "Speaker" + - "Headset Mic" + - "Internal Mic 1" + - "Internal Mic 2" + + # CODEC Pins + - IN1P + - IN1N + - IN2P + - IN2N + - MICBIAS1 + - DMIC1 + - DMIC2 + - DMIC3 + - DMIC4 + - "DMIC L1" + - "DMIC L2" + - "DMIC L3" + - "DMIC L4" + - "DMIC R1" + - "DMIC R2" + - "DMIC R3" + - "DMIC R4" + - LOUT1 + - LOUT2 + - LOUT3 + - PDM1L + - PDM1R + - PDM2L + - PDM2R + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, + <&clk 217>, + <&clk 121>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml new file mode 100644 index 000000000..943e7c017 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Line In Jack" + - "Mic Jack" + + # CODEC Pins + - HP_OUT + - LINE_OUT + - LINE_IN + - MIC_IN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30 SGTL5000"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml new file mode 100644 index 000000000..8c87cd166 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + const: nvidia,tegra-audio-trimslice + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-trimslice"; + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml new file mode 100644 index 000000000..a5b431d7d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Mic Jack" + + # CODEC Pins + - LOUT1 + - LOUT2 + - ROUT1 + - ROUT2 + - MONO1 + - MONO2 + - OUT3 + - OUT4 + - LINE1 + - LINE2 + - RXP + - RXN + - ACIN + - ACOP + - MIC1N + - MIC1 + - MIC2N + - MIC2 + - "Mic Bias" + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8753-whistler", + "nvidia,tegra-audio-wm8753"; + nvidia,model = "tegra-wm8753-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "LOUT1", + "Headphone Jack", "ROUT1"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8753>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml new file mode 100644 index 000000000..1b836acab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Int Spk" + - "Mic Jack" + - "Int Mic" + + # CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - DMICDAT + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - LOP + - LON + - ROP + - RON + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8903-harmony", + "nvidia,tegra-audio-wm8903"; + nvidia,model = "tegra-wm8903-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "HPOUTR", + "Headphone Jack", "HPOUTL", + "Int Spk", "ROP", + "Int Spk", "RON", + "Int Spk", "LOP", + "Int Spk", "LON", + "Mic Jack", "MICBIAS", + "IN1L", "Mic Jack"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8903>; + + nvidia,spkr-en-gpios = <&codec 2 0>; + nvidia,hp-det-gpios = <&gpio 178 0>; + nvidia,int-mic-en-gpios = <&gpio 184 0>; + nvidia,ext-mic-en-gpios = <&gpio 185 0>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml new file mode 100644 index 000000000..a14482833 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone" + - "LineIn" + - "Mic" + + # CODEC Pins + - MONOOUT + - HPOUTL + - HPOUTR + - LOUT2 + - ROUT2 + - OUT3 + - LINEINL + - LINEINR + - PHONE + - PCBEEP + - MIC1 + - MIC2 + - "Mic Bias" + +required: + - nvidia,ac97-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm9712-colibri_t20", + "nvidia,tegra-audio-wm9712"; + nvidia,model = "Toradex Colibri T20"; + + nvidia,audio-routing = + "Headphone", "HPOUTL", + "Headphone", "HPOUTR", + "LineIn", "LINEINL", + "LineIn", "LINEINR", + "Mic", "MIC1"; + + nvidia,ac97-controller = <&ac97>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml new file mode 100644 index 000000000..d82415c21 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^asrc@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra186-asrc + - items: + - enum: + - nvidia,tegra234-asrc + - nvidia,tegra194-asrc + - const: nvidia,tegra186-asrc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^ASRC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + ASRC has seven input ports and six output ports. Accordingly ACIF + (Audio Client Interfaces) port nodes are defined to represent the + ASRC inputs (port 0 to 6) and outputs (port 7 to 12). These are + connected to corresponding ports on AHUB (Audio Hub). Additional + input (port 6) is for receiving ratio information from estimator. + + patternProperties: + '^port@[0-6]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF input ports + '^port@[7-9]|1[1-2]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + asrc@2910000 { + compatible = "nvidia,tegra186-asrc"; + reg = <0x2910000 0x2000>; + sound-name-prefix = "ASRC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml new file mode 100644 index 000000000..3d538df87 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^dspk@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra186-dspk + - items: + - enum: + - nvidia,tegra234-dspk + - nvidia,tegra194-dspk + - const: nvidia,tegra186-dspk + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: dspk + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + sound-name-prefix: + pattern: "^DSPK[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DSPK ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DSPK DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + - sound-name-prefix + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra186-clock.h> + + dspk@2905000 { + compatible = "nvidia,tegra186-dspk"; + reg = <0x2905000 0x100>; + clocks = <&bpmp TEGRA186_CLK_DSPK1>; + clock-names = "dspk"; + assigned-clocks = <&bpmp TEGRA186_CLK_DSPK1>; + assigned-clock-parents = <&bpmp TEGRA186_CLK_PLL_A_OUT0>; + assigned-clock-rates = <12288000>; + sound-name-prefix = "DSPK1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt new file mode 100644 index 000000000..eaf00102d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt @@ -0,0 +1,36 @@ +NVIDIA Tegra 20 AC97 controller + +Required properties: +- compatible : "nvidia,tegra20-ac97" +- reg : Should contain AC97 controller registers location and length +- interrupts : Should contain AC97 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: + - ac97 +- 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. +- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number + of the GPIO used to reset the external AC97 codec +- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number + of the GPIO corresponding with the AC97 DAP _FS line + +Example: + +ac97@70002000 { + compatible = "nvidia,tegra20-ac97"; + reg = <0x70002000 0x200>; + interrupts = <0 81 0x04>; + nvidia,codec-reset-gpio = <&gpio 170 0>; + nvidia,codec-sync-gpio = <&gpio 120 0>; + clocks = <&tegra_car 3>; + resets = <&tegra_car 3>; + reset-names = "ac97"; + dmas = <&apbdma 12>, <&apbdma 12>; + dma-names = "rx", "tx"; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt new file mode 100644 index 000000000..6de3a7ee4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt @@ -0,0 +1,12 @@ +NVIDIA Tegra 20 DAS (Digital Audio Switch) controller + +Required properties: +- compatible : "nvidia,tegra20-das" +- reg : Should contain DAS registers location and length + +Example: + +das@70000c00 { + compatible = "nvidia,tegra20-das"; + reg = <0x70000c00 0x80>; +}; 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 000000000..68ae124ea --- /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# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 I2S Controller + +description: | + The I2S Controller streams synchronous serial audio data between system + memory and an external audio device. The controller supports the I2S Left + Justified Mode, Right Justified Mode, and DSP mode formats. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-i2s + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: i2s + + interrupts: + maxItems: 1 + + clocks: + minItems: 1 + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = <45>; + clocks = <&tegra_car 11>; + resets = <&tegra_car 11>; + reset-names = "i2s"; + dmas = <&apbdma 21>, <&apbdma 21>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml new file mode 100644 index 000000000..60a368a13 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 S/PDIF Controller + +description: | + The S/PDIF controller supports both input and output in serial audio + digital interface format. The input controller can digitally recover + a clock from the received stream. The S/PDIF controller is also used + to generate the embedded audio for HDMI output channel. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 2 + + clock-names: + items: + - const: out + - const: in + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + spdif@70002400 { + compatible = "nvidia,tegra20-spdif"; + reg = <0x70002400 0x200>; + interrupts = <77>; + clocks = <&clk 99>, <&clk 98>; + clock-names = "out", "in"; + resets = <&rst 10>; + dmas = <&apbdma 3>, <&apbdma 3>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml new file mode 100644 index 000000000..15ab40aea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml @@ -0,0 +1,129 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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)$" + description: + Should be "rx1", "rx2" ... "rx20" for DMA Rx channel + Should be "tx1", "tx2" ... "tx20" for DMA Tx channel + minItems: 1 + maxItems: 40 + +required: + - compatible + - reg + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml new file mode 100644 index 000000000..ea0dc0ece --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^adx@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-adx + - items: + - enum: + - nvidia,tegra234-adx + - nvidia,tegra194-adx + - nvidia,tegra186-adx + - const: nvidia,tegra210-adx + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^ADX[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + ADX has one input and four outputs. Accordingly ACIF (Audio Client + Interface) port nodes are defined to represent ADX input (port 0) + and outputs (ports 1 to 4). These are connected to corresponding + ports on AHUB (Audio Hub). + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF input port + patternProperties: + '^port@[1-4]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + adx@702d3800 { + compatible = "nvidia,tegra210-adx"; + reg = <0x702d3800 0x100>; + sound-name-prefix = "ADX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml new file mode 100644 index 000000000..89f7805de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml @@ -0,0 +1,196 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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": + const: 1 + + "#size-cells": + const: 1 + + ranges: true + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Contains list of ACIF (Audio CIF) port nodes for AHUB (Audio Hub). + These are connected to ACIF interfaces of AHUB clients. Thus the + number of port nodes depend on the number of clients that AHUB may + have depending on the SoC revision. + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +patternProperties: + '^i2s@[0-9a-f]+$': + type: object + + '^dmic@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-dmic.yaml# + + '^admaif@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-admaif.yaml# + + '^dspk@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-dspk.yaml# + + '^mvc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mvc.yaml# + + '^sfc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-sfc.yaml# + + '^amx@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-amx.yaml# + + '^adx@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-adx.yaml# + + '^amixer@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mixer.yaml# + + '^asrc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-asrc.yaml# + + '^processing-engine@[0-9a-f]+$': + 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 000000000..1aff61f07 --- /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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^amx@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-amx + - items: + - const: nvidia,tegra186-amx + - const: nvidia,tegra210-amx + - const: nvidia,tegra194-amx + - items: + - const: nvidia,tegra234-amx + - const: nvidia,tegra194-amx + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^AMX[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + AMX has four inputs and one output. Accordingly ACIF (Audio Client + Interfaces) port nodes are defined to represent AMX inputs (port 0 + to 3) and output (port 4). These are connected to corresponding + ports on AHUB (Audio Hub). + + patternProperties: + '^port@[0-3]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF input ports + + properties: + port@4: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF output port + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amx@702d3000 { + compatible = "nvidia,tegra210-amx"; + reg = <0x702d3000 0x100>; + sound-name-prefix = "AMX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml new file mode 100644 index 000000000..0f9d2b461 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^dmic@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-dmic + - items: + - enum: + - nvidia,tegra234-dmic + - nvidia,tegra194-dmic + - nvidia,tegra186-dmic + - const: nvidia,tegra210-dmic + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: dmic + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + sound-name-prefix: + pattern: "^DMIC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC DAP (Digital Audio Port) interface which can be connected + to external audio codec for capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + dmic@702d4000 { + compatible = "nvidia,tegra210-dmic"; + reg = <0x702d4000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + clock-names = "dmic"; + assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <3072000>; + sound-name-prefix = "DMIC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml new file mode 100644 index 000000000..12cd17eed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^i2s@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-i2s + - items: + - enum: + - nvidia,tegra234-i2s + - nvidia,tegra194-i2s + - nvidia,tegra186-i2s + - const: nvidia,tegra210-i2s + + reg: + maxItems: 1 + + clocks: + minItems: 1 + items: + - description: I2S bit clock + - description: + Sync input clock, which can act as clock source to other I/O + modules in AHUB. The Tegra I2S driver sets this clock rate as + per bit clock rate. I/O module which wants to use this clock + as source, can mention this clock as parent in the DT bindings. + This is an optional clock entry, since it is only required when + some other I/O wants to reference from a particular I2Sx + instance. + + clock-names: + minItems: 1 + items: + - const: i2s + - const: sync_input + + assigned-clocks: + minItems: 1 + maxItems: 2 + + assigned-clock-parents: + minItems: 1 + maxItems: 2 + + assigned-clock-rates: + minItems: 1 + maxItems: 2 + + sound-name-prefix: + pattern: "^I2S[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback or capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + reg = <0x702d1000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + sound-name-prefix = "I2S1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml new file mode 100644 index 000000000..5b9198602 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mbdrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 MBDRC + +description: + The Multi Band Dynamic Range Compressor (MBDRC) is part of Output + Processing Engine (OPE) which interfaces with Audio Hub (AHUB) via + Audio Client Interface (ACIF). MBDRC can be used as a traditional + single full band or a dual band or a multi band dynamic processor. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-mbdrc + - items: + - enum: + - nvidia,tegra234-mbdrc + - nvidia,tegra194-mbdrc + - nvidia,tegra186-mbdrc + - const: nvidia,tegra210-mbdrc + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + dynamic-range-compressor@702d8200 { + compatible = "nvidia,tegra210-mbdrc"; + reg = <0x702d8200 0x200>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml new file mode 100644 index 000000000..570b03282 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^amixer@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-amixer + - items: + - enum: + - nvidia,tegra234-amixer + - nvidia,tegra194-amixer + - nvidia,tegra186-amixer + - const: nvidia,tegra210-amixer + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^MIXER[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Mixer has ten inputs and five outputs. Accordingly ACIF (Audio + Client Interfaces) port nodes are defined to represent Mixer + inputs (port 0 to 9) and outputs (port 10 to 14). These are + connected to corresponding ports on AHUB (Audio Hub). + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF input ports + '^port@[10-14]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amixer@702dbb00 { + compatible = "nvidia,tegra210-amixer"; + reg = <0x702dbb00 0x800>; + sound-name-prefix = "MIXER1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml new file mode 100644 index 000000000..4aecbc847 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^mvc@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-mvc + - items: + - enum: + - nvidia,tegra234-mvc + - nvidia,tegra194-mvc + - nvidia,tegra186-mvc + - const: nvidia,tegra210-mvc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^MVC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + mvc@702da000 { + compatible = "nvidia,tegra210-mvc"; + reg = <0x702da000 0x200>; + sound-name-prefix = "MVC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml new file mode 100644 index 000000000..9dc9ba590 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + - $ref: name-prefix.yaml# + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-ope + - items: + - enum: + - nvidia,tegra234-ope + - nvidia,tegra194-ope + - nvidia,tegra186-ope + - const: nvidia,tegra210-ope + + reg: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + ranges: true + + sound-name-prefix: + pattern: "^OPE[1-9]$" + + 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 + $ref: nvidia,tegra210-mbdrc.yaml# + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + processing-engine@702d8000 { + compatible = "nvidia,tegra210-ope"; + reg = <0x702d8000 0x100>; + sound-name-prefix = "OPE1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml new file mode 100644 index 000000000..1e373c49d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-peq.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 PEQ + +description: + The Parametric Equalizer (PEQ) is a cascade of biquad filters with + each filter tuned based on certain parameters. It can be used to + equalize the irregularities in the speaker frequency response. + PEQ sits inside Output Processing Engine (OPE) which interfaces + with Audio Hub (AHUB) via Audio Client Interface (ACIF). + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-peq + - items: + - enum: + - nvidia,tegra234-peq + - nvidia,tegra194-peq + - nvidia,tegra186-peq + - const: nvidia,tegra210-peq + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + equalizer@702d8100 { + compatible = "nvidia,tegra210-peq"; + reg = <0x702d8100 0x100>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml new file mode 100644 index 000000000..694f890d6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: name-prefix.yaml# + +properties: + $nodename: + pattern: "^sfc@[0-9a-f]*$" + + compatible: + oneOf: + - const: nvidia,tegra210-sfc + - items: + - enum: + - nvidia,tegra234-sfc + - nvidia,tegra194-sfc + - nvidia,tegra186-sfc + - const: nvidia,tegra210-sfc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^SFC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + SFC ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + SFC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + sfc@702d2000 { + compatible = "nvidia,tegra210-sfc"; + reg = <0x702d2000 0x200>; + sound-name-prefix = "SFC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt new file mode 100644 index 000000000..0e9a1895d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -0,0 +1,88 @@ +NVIDIA Tegra30 AHUB (Audio Hub) + +Required properties: +- compatible : For Tegra30, must contain "nvidia,tegra30-ahub". For Tegra114, + must contain "nvidia,tegra114-ahub". For Tegra124, must contain + "nvidia,tegra124-ahub". Otherwise, must contain "nvidia,<chip>-ahub", + plus at least one of the above, where <chip> is tegra132. +- reg : Should contain the register physical address and length for each of + the AHUB's register blocks. + - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks. + - Tegra114 requires an additional entry, for the APBIF2 register block. +- interrupts : Should contain AHUB interrupt +- 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: + - d_audio + - apbif +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: + Tegra30 and later: + - d_audio + - apbif + - i2s0 + - i2s1 + - i2s2 + - i2s3 + - i2s4 + - dam0 + - dam1 + - dam2 + - spdif + Tegra114 and later additionally require: + - amx + - adx + Tegra124 and later additionally require: + - amx1 + - adx1 + - afc0 + - afc1 + - afc2 + - afc3 + - afc4 + - afc5 +- ranges : The bus address mapping for the configlink register bus. + Can be empty since the mapping is 1:1. +- dmas : Must contain an entry for each entry in clock-names. + See ../dma/dma.txt for details. +- dma-names : Must include the following entries: + - rx0 .. rx<n> + - tx0 .. tx<n> + ... where n is: + Tegra30: 3 + Tegra114, Tegra124: 9 +- #address-cells : For the configlink bus. Should be <1>; +- #size-cells : For the configlink bus. Should be <1>. + +AHUB client modules need to specify the IDs of their CIFs (Client InterFaces). +For RX CIFs, the numbers indicate the register number within AHUB routing +register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1). +For TX CIFs, the numbers indicate the bit position within the AHUB routing +registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1). + +Example: + +ahub@70080000 { + compatible = "nvidia,tegra30-ahub"; + reg = <0x70080000 0x200 0x70080200 0x100>; + interrupts = < 0 103 0x04 >; + nvidia,dma-request-selector = <&apbdma 1>; + clocks = <&tegra_car 106>, <&tegra_car 107>; + clock-names = "d_audio", "apbif"; + resets = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>, + <&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>, + <&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>, + <&tegra_car 110>, <&tegra_car 10>; + reset-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2", + "i2s3", "i2s4", "dam0", "dam1", "dam2", + "spdif"; + dmas = <&apbdma 1>, <&apbdma 1>; + <&apbdma 2>, <&apbdma 2>; + <&apbdma 3>, <&apbdma 3>; + <&apbdma 4>, <&apbdma 4>; + dma-names = "rx0", "tx0", "rx1", "tx1", "rx2", "tx2", "rx3", "tx3"; + ranges; + #address-cells = <1>; + #size-cells = <1>; +}; 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 000000000..12c31b4b9 --- /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/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt new file mode 100644 index 000000000..38caa936f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt @@ -0,0 +1,27 @@ +NVIDIA Tegra30 I2S controller + +Required properties: +- compatible : For Tegra30, must contain "nvidia,tegra30-i2s". For Tegra124, + must contain "nvidia,tegra124-i2s". Otherwise, must contain + "nvidia,<chip>-i2s" plus at least one of the above, where <chip> is + tegra114 or tegra132. +- reg : Should contain I2S registers location and length +- clocks : Must contain one entry, for the module clock. + See ../clocks/clock-bindings.txt for details. +- 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 +- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback) + first, tx (capture) second. See nvidia,tegra30-ahub.txt for values. + +Example: + +i2s@70080300 { + compatible = "nvidia,tegra30-i2s"; + reg = <0x70080300 0x100>; + nvidia,ahub-cif-ids = <4 4>; + clocks = <&tegra_car 11>; + resets = <&tegra_car 11>; + reset-names = "i2s"; +}; diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml new file mode 100644 index 000000000..7f2e68ff6 --- /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: name-prefix.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/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 000000000..462b04e82 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of 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. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack-detection: Need to be present if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-dmic.txt b/Documentation/devicetree/bindings/sound/omap-dmic.txt new file mode 100644 index 000000000..418e30e72 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-dmic.txt @@ -0,0 +1,20 @@ +* Texas Instruments OMAP4+ Digital Microphone Module + +Required properties: +- compatible: "ti,omap4-dmic" +- reg: Register location and size as an array: + <MPU access base address, size>, + <L3 interconnect address, size>; +- interrupts: Interrupt number for DMIC +- ti,hwmods: Name of the hwmod associated with OMAP dmic IP + +Example: + +dmic: dmic@4012e000 { + compatible = "ti,omap4-dmic"; + reg = <0x4012e000 0x7f>, /* MPU private access */ + <0x4902e000 0x7f>; /* L3 Interconnect */ + interrupts = <0 114 0x4>; + interrupt-parent = <&gic>; + ti,hwmods = "dmic"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt new file mode 100644 index 000000000..ae8bf703c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -0,0 +1,36 @@ +* Texas Instruments OMAP2+ McBSP module + +Required properties: +- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420 + "ti,omap2430-mcbsp" for McBSP on OMAP2430 + "ti,omap3-mcbsp" for McBSP on OMAP3 + "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC +- reg: Register location and size, for OMAP4+ as an array: + <MPU access base address, size>, + <L3 interconnect address, size>; +- reg-names: Array of strings associated with the address space +- interrupts: Interrupt numbers for the McBSP port, as an array in case the + McBSP IP have more interrupt lines: + <OCP compliant irq>, + <TX irq>, + <RX irq>; +- interrupt-names: Array of strings associated with the interrupt numbers +- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) +- ti,hwmods: Name of the hwmod associated to the McBSP port + +Example: + +mcbsp2: mcbsp@49022000 { + compatible = "ti,omap3-mcbsp"; + reg = <0x49022000 0xff>, + <0x49028000 0xff>; + reg-names = "mpu", "sidetone"; + interrupts = <0 17 0x4>, /* OCP compliant interrupt */ + <0 62 0x4>, /* TX interrupt */ + <0 63 0x4>, /* RX interrupt */ + <0 4 0x4>; /* Sidetone */ + interrupt-names = "common", "tx", "rx", "sidetone"; + interrupt-parent = <&intc>; + ti,buffer-size = <1280>; + ti,hwmods = "mcbsp2"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt new file mode 100644 index 000000000..ff98a0cb5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt @@ -0,0 +1,30 @@ +* Texas Instruments OMAP4+ McPDM + +Required properties: +- compatible: "ti,omap4-mcpdm" +- reg: Register location and size as an array: + <MPU access base address, size>, + <L3 interconnect address, size>; +- interrupts: Interrupt number for McPDM +- ti,hwmods: Name of the hwmod associated to the McPDM +- clocks: phandle for the pdmclk provider, likely <&twl6040> +- clock-names: Must be "pdmclk" + +Example: + +mcpdm: mcpdm@40132000 { + compatible = "ti,omap4-mcpdm"; + reg = <0x40132000 0x7f>, /* MPU private access */ + <0x49032000 0x7f>; /* L3 Interconnect */ + interrupts = <0 112 0x4>; + interrupt-parent = <&gic>; + ti,hwmods = "mcpdm"; +}; + +In board DTS file the pdmclk needs to be added: + +&mcpdm { + clocks = <&twl6040>; + clock-names = "pdmclk"; + status = "okay"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt new file mode 100644 index 000000000..f6a715e4e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt @@ -0,0 +1,62 @@ +* Texas Instruments SoC with twl4030 based audio setups + +Required properties: +- compatible: "ti,omap-twl4030" +- ti,model: Name of the sound card (for example "omap3beagle") +- ti,mcbsp: phandle for the McBSP node + +Optional properties: +- ti,codec: phandle for the twl4030 audio node +- ti,mcbsp-voice: phandle for the McBSP node connected to the voice port of twl +- ti, jack-det-gpio: Jack detect GPIO +- ti,audio-routing: List of 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. + If the routing is not provided all possible connection will be available + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earpiece Spk + * Handsfree Spk + * Ext Spk + * Main Mic + * Sub Mic + * Headset Mic + * Carkit Mic + * Digital0 Mic + * Digital1 Mic + * Line In + +twl4030 pins: + * HSOL + * HSOR + * EARPIECE + * HFL + * HFR + * PREDRIVEL + * PREDRIVER + * CARKITL + * CARKITR + * MAINMIC + * SUBMIC + * HSMIC + * DIGIMIC0 + * DIGIMIC1 + * CARKITMIC + * AUXL + * AUXR + + * Headset Mic Bias + * Mic Bias 1 /* Used for Main Mic or Digimic0 */ + * Mic Bias 2 /* Used for Sub Mic or Digimic1 */ + +Example: + +sound { + compatible = "ti,omap-twl4030"; + ti,model = "omap3beagle"; + + ti,mcbsp = <&mcbsp2>; +}; diff --git a/Documentation/devicetree/bindings/sound/pcm1789.txt b/Documentation/devicetree/bindings/sound/pcm1789.txt new file mode 100644 index 000000000..3c74ed220 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm1789.txt @@ -0,0 +1,22 @@ +Texas Instruments pcm1789 DT bindings + +PCM1789 is a simple audio codec that can be connected via +I2C or SPI. Currently, only I2C bus is supported. + +Required properties: + + - compatible: "ti,pcm1789" + +Required properties on I2C: + + - reg: the I2C address + - reset-gpios: GPIO to control the RESET pin + +Examples: + + audio-codec@4c { + compatible = "ti,pcm1789"; + reg = <0x4c>; + reset-gpios = <&gpio2 14 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm179x.txt b/Documentation/devicetree/bindings/sound/pcm179x.txt new file mode 100644 index 000000000..436c2b247 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm179x.txt @@ -0,0 +1,27 @@ +Texas Instruments pcm179x DT bindings + +This driver supports both the I2C and SPI bus. + +Required properties: + + - compatible: "ti,pcm1792a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the I2C address + + +Examples: + + codec_spi: 1792a@0 { + compatible = "ti,pcm1792a"; + spi-max-frequency = <600000>; + }; + + codec_i2c: 1792a@4c { + compatible = "ti,pcm1792a"; + reg = <0x4c>; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm186x.txt b/Documentation/devicetree/bindings/sound/pcm186x.txt new file mode 100644 index 000000000..1087f4855 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm186x.txt @@ -0,0 +1,42 @@ +Texas Instruments PCM186x Universal Audio ADC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "ti,pcm1862", + "ti,pcm1863", + "ti,pcm1864", + "ti,pcm1865" + + - reg : The I2C address of the device for I2C, the chip select + number for SPI. + + - avdd-supply: Analog core power supply (3.3v) + - dvdd-supply: Digital core power supply + - iovdd-supply: Digital IO power supply + See regulator/regulator.txt for more information + +CODEC input pins: + * VINL1 + * VINR1 + * VINL2 + * VINR2 + * VINL3 + * VINR3 + * VINL4 + * VINR4 + +The pins can be used in referring sound node's audio-routing property. + +Example: + + pcm186x: audio-codec@4a { + compatible = "ti,pcm1865"; + reg = <0x4a>; + + avdd-supply = <®_3v3_analog>; + dvdd-supply = <®_3v3>; + iovdd-supply = <®_1v8>; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt new file mode 100644 index 000000000..97de66932 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -0,0 +1,23 @@ +PCM3060 audio CODEC + +This driver supports both I2C and SPI. + +Required properties: + +- compatible: "ti,pcm3060" + +- reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + +Examples: + + pcm3060: pcm3060@46 { + compatible = "ti,pcm3060"; + reg = <0x46>; + ti,out-single-ended = "true"; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt new file mode 100644 index 000000000..c63ab0b6e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODECs + +These devices does not use I2C or SPI. + +Required properties: + + - compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt new file mode 100644 index 000000000..3aae3b41b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -0,0 +1,52 @@ +PCM512x audio CODECs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or + "ti,pcm5142" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the + device, as covered in bindings/regulator/regulator.txt + +Optional properties: + + - clocks : A clock specifier for the clock connected as SCLK. If this + is absent the device will be configured to clock from BCLK. If pll-in + and pll-out are specified in addition to a clock, the device is + configured to accept clock input on a specified gpio pin. + + - pll-in, pll-out : gpio pins used to connect the pll using <1> + 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. + +Examples: + + pcm5122: pcm5122@4c { + compatible = "ti,pcm5122"; + reg = <0x4c>; + + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + }; + + + pcm5142: pcm5142@4c { + compatible = "ti,pcm5142"; + reg = <0x4c>; + + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + + clocks = <&sck>; + pll-in = <3>; + pll-out = <6>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt new file mode 100644 index 000000000..e1b9fa8a5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt @@ -0,0 +1,128 @@ +* Qualcomm Technologies APQ8096 ASoC sound card driver + +This binding describes the APQ8096 sound card, which uses qdsp for audio. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,apq8096-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: + Valid names include: + + Board Connectors: + "Headphone Left" + "Headphone Right" + "Earphone" + "Line Out1" + "Line Out2" + "Line Out3" + "Line Out4" + "Analog Mic1" + "Analog Mic2" + "Analog Mic3" + "Analog Mic4" + "Analog Mic5" + "Analog Mic6" + "Digital Mic2" + "Digital Mic3" + + Audio pins and MicBias on WCD9335 Codec: + "MIC_BIAS1" + "MIC_BIAS2" + "MIC_BIAS3" + "MIC_BIAS4" + "AMIC1" + "AMIC2" + "AMIC3" + "AMIC4" + "AMIC5" + "AMIC6" + "AMIC6" + "DMIC1" + "DMIC2" + "DMIC3" + +- model: + Usage: required + 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. + +- 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: Optional + 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 with arguments> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. + +Obsolete: + qcom,model: String for soundcard name (Use model instead) + qcom,audio-routing: A list of the connections between audio components. + (Use audio-routing instead) + +Example: + +audio { + compatible = "qcom,apq8096-sndcard"; + model = "DB820c"; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + hdmi-dai-link { + link-name = "HDMI Playback"; + cpu { + sound-dai = <&q6afe HDMI_RX>; + }; + + platform { + sound-dai = <&q6adm>; + }; + + codec { + sound-dai = <&hdmi 0>; + }; + }; +}; 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 000000000..ef18a572a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -0,0 +1,293 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-cpu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. LPASS CPU dai driver bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + - Rohit kumar <rohitkr@codeaurora.org> + +description: | + Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist + of MI2S interface for audio data transfer on external codecs. LPASS cpu driver + is a module to configure Low-Power Audio Interface(LPAIF) core registers + across different IP versions. + +properties: + compatible: + enum: + - qcom,lpass-cpu + - qcom,apq8016-lpass-cpu + - qcom,sc7180-lpass-cpu + - qcom,sc7280-lpass-cpu + + reg: + minItems: 1 + maxItems: 6 + description: LPAIF core registers + + reg-names: + minItems: 1 + maxItems: 6 + + clocks: + minItems: 3 + maxItems: 7 + + clock-names: + minItems: 1 + maxItems: 10 + + interrupts: + minItems: 1 + maxItems: 4 + description: LPAIF DMA buffer interrupt + + interrupt-names: + minItems: 1 + maxItems: 4 + + qcom,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle for the audio DSP node + + iommus: + minItems: 2 + maxItems: 3 + description: Phandle to apps_smmu node with sid mask + + power-domains: + maxItems: 1 + + power-domain-names: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]$": + type: object + description: | + LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node + depends on the specific driver providing the functionality and + properties. + properties: + reg: + maxItems: 1 + description: Must be one of the DAI ID + + qcom,playback-sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: list of MI2S data lines for playback + + qcom,capture-sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: list of MI2S data lines for capture + + required: + - reg + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - interrupts + - interrupt-names + - '#sound-dai-cells' + +additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: qcom,lpass-cpu + + then: + properties: + clock-names: + items: + - const: ahbix-clk + - const: mi2s-osr-clk + - const: mi2s-bit-clk + + - if: + properties: + compatible: + contains: + const: qcom,apq8016-lpass-cpu + + then: + properties: + clock-names: + items: + - const: ahbix-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + - const: mi2s-bit-clk2 + - const: mi2s-bit-clk3 + - const: pcnoc-mport-clk + - const: pcnoc-sway-clk + + - if: + properties: + compatible: + contains: + const: qcom,sc7180-lpass-cpu + + then: + properties: + clock-names: + oneOf: + - items: #for I2S + - const: pcnoc-sway-clk + - const: audio-core + - const: mclk0 + - const: pcnoc-mport-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + - items: #for HDMI + - const: pcnoc-sway-clk + - const: audio-core + - const: pcnoc-mport-clk + reg-names: + anyOf: + - items: #for I2S + - const: lpass-lpaif + - items: #for I2S and HDMI + - const: lpass-hdmiif + - const: lpass-lpaif + interrupt-names: + anyOf: + - items: #for I2S + - const: lpass-irq-lpaif + - items: #for I2S and HDMI + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + required: + - iommus + - power-domains + + - if: + properties: + compatible: + contains: + const: qcom,sc7280-lpass-cpu + + then: + properties: + clock-names: + oneOf: + - items: #for I2S + - const: aon_cc_audio_hm_h + - const: audio_cc_ext_mclk0 + - const: core_cc_sysnoc_mport_core + - const: core_cc_ext_if0_ibit + - const: core_cc_ext_if1_ibit + - items: #for Soundwire + - const: aon_cc_audio_hm_h + - const: audio_cc_codec_mem + - const: audio_cc_codec_mem0 + - const: audio_cc_codec_mem1 + - const: audio_cc_codec_mem2 + - const: aon_cc_va_mem0 + - items: #for HDMI + - const: core_cc_sysnoc_mport_core + + reg-names: + anyOf: + - items: #for I2S + - const: lpass-lpaif + - items: #for I2S and HDMI + - const: lpass-hdmiif + - const: lpass-lpaif + - items: #for I2S, soundwire and HDMI + - const: lpass-hdmiif + - const: lpass-lpaif + - const: lpass-rxtx-cdc-dma-lpm + - const: lpass-rxtx-lpaif + - const: lpass-va-lpaif + - const: lpass-va-cdc-dma-lpm + interrupt-names: + anyOf: + - items: #for I2S + - const: lpass-irq-lpaif + - items: #for I2S and HDMI + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - items: #for I2S, soundwire and HDMI + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - const: lpass-irq-vaif + - const: lpass-irq-rxtxif + power-domain-names: + allOf: + - items: + - const: lcx + + required: + - iommus + - power-domains + +examples: + - | + #include <dt-bindings/sound/sc7180-lpass.h> + + soc { + #address-cells = <2>; + #size-cells = <2>; + lpass@62d80000 { + compatible = "qcom,sc7180-lpass-cpu"; + + reg = <0 0x62d87000 0 0x68000>, + <0 0x62f00000 0 0x29000>; + reg-names = "lpass-hdmiif", + "lpass-lpaif"; + iommus = <&apps_smmu 0x1020 0>, + <&apps_smmu 0x1032 0>; + power-domains = <&lpass_hm 0>; + + clocks = <&gcc 131>, + <&lpasscorecc 6>, + <&lpasscorecc 7>, + <&lpasscorecc 10>, + <&lpasscorecc 8>, + <&lpasscorecc 9>; + + clock-names = "pcnoc-sway-clk", "audio-core", + "mclk0", "pcnoc-mport-clk", + "mi2s-bit-clk0", "mi2s-bit-clk1"; + + interrupts = <0 160 1>, + <0 268 1>; + interrupt-names = "lpass-irq-lpaif", + "lpass-irq-hdmi"; + #sound-dai-cells = <1>; + + #address-cells = <1>; + #size-cells = <0>; + /* Optional to set different MI2S SD lines */ + dai-link@0 { + reg = <MI2S_PRIMARY>; + qcom,playback-sd-lines = <1>; + qcom,capture-sd-lines = <0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml new file mode 100644 index 000000000..c8d803097 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-rx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) RX Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + - qcom,sc8280xp-lpass-rx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 3 + maxItems: 5 + + clock-names: + oneOf: + - items: #for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: #for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + clock-output-names: + items: + - const: mclk + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3200000 { + compatible = "qcom,sm8250-lpass-rx-macro"; + reg = <0x3200000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 0>, + <&audiocc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml new file mode 100644 index 000000000..de8297b35 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-tx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) TX Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + - qcom,sm8450-lpass-tx-macro + - qcom,sc8280xp-lpass-tx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 5 + + clock-names: + oneOf: + - items: #for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: #for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + clock-output-names: + items: + - const: mclk + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3220000 { + compatible = "qcom,sm8250-lpass-tx-macro"; + reg = <0x3220000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&aoncc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + qcom,dmic-sample-rate = <600000>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml new file mode 100644 index 000000000..9f473c08c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-va-macro + - qcom,sm8250-lpass-va-macro + - qcom,sm8450-lpass-va-macro + - qcom,sc8280xp-lpass-va-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 3 + + clock-names: + oneOf: + - items: #for ADSP based platforms + - const: mclk + - const: core + - const: dcodec + - items: #for ADSP bypass based platforms + - const: mclk + + clock-output-names: + items: + - const: fsgen + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + + vdd-micb-supply: + description: phandle to voltage regulator of MIC Bias + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3370000 { + compatible = "qcom,sm8250-lpass-va-macro"; + reg = <0x3370000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; + clock-names = "mclk", "core", "dcodec"; + clock-output-names = "fsgen"; + qcom,dmic-sample-rate = <600000>; + vdd-micb-supply = <&vreg_s4a_1p8>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml new file mode 100644 index 000000000..4959ad658 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-wsa-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec DT bindings + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + maxItems: 5 + + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + clock-output-names: + items: + - const: mclk + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + + vdd-micb-supply: + description: phandle to voltage regulator of MIC Bias + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3240000 { + compatible = "qcom,sm8250-lpass-wsa-macro"; + reg = <0x3240000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 1>, + <&audiocc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt new file mode 100644 index 000000000..e7d17dda5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt @@ -0,0 +1,101 @@ +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,msm8916-wcd-digital.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt new file mode 100644 index 000000000..1c8e4cb25 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt @@ -0,0 +1,20 @@ +msm8916 digital audio CODEC + +## Bindings for codec core in lpass: + +Required properties + - compatible = "qcom,msm8916-wcd-digital-codec"; + - reg: address space for lpass codec. + - clocks: Handle to mclk and ahbclk + - clock-names: should be "mclk", "ahbix-clk". + +Example: + +audio-codec@771c000{ + compatible = "qcom,msm8916-wcd-digital-codec"; + reg = <0x0771c000 0x400>; + clocks = <&gcc GCC_ULTAUDIO_AHBFABRIC_IXFABRIC_CLK>, + <&gcc GCC_CODEC_DIGCODEC_CLK>; + clock-names = "ahbix-clk", "mclk"; + #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 000000000..d0f7a79e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6adm-routing.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Device Manager (Q6ADM) routing + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Qualcomm Audio Device Manager (Q6ADM) routing node represents routing + specific configuration. + +properties: + compatible: + enum: + - qcom,q6adm-routing + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6asm.h> + + apr { + compatible = "qcom,apr-v2"; + qcom,domain = <APR_DOMAIN_ADSP>; + #address-cells = <1>; + #size-cells = <0>; + + service@8 { + compatible = "qcom,q6adm"; + reg = <APR_SVC_ADM>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + routing { + compatible = "qcom,q6adm-routing"; + #sound-dai-cells = <0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml new file mode 100644 index 000000000..24f7bf2bf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml @@ -0,0 +1,46 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/qcom,q6apm-dai.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Qualcomm Audio Process Manager Digital Audio Interfaces binding + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm APM DAIs in DSP + +properties: + compatible: + const: qcom,q6apm-dais + + iommus: + maxItems: 1 + +required: + - compatible + - iommus + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,gpr.h> + gpr { + compatible = "qcom,gpr"; + #address-cells = <1>; + #size-cells = <0>; + qcom,domain = <GPR_DOMAIN_ID_ADSP>; + + service@1 { + compatible = "qcom,q6apm"; + reg = <1>; + + dais { + compatible = "qcom,q6apm-dais"; + iommus = <&apps_smmu 0x1801 0x0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml b/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml new file mode 100644 index 000000000..8deb8ffb1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml @@ -0,0 +1,112 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: | + The direction of the dai stream:: + - Q6ASM_DAI_TX_RX (0) for both tx and rx + - Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + - Q6ASM_DAI_RX (2) for only rx (Playback/Decode) + + is-compress-dai: + type: boolean + description: + Compress offload dai. + + dependencies: + is-compress-dai: ["direction"] + + required: + - reg + + additionalProperties: false + +required: + - compatible + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6asm.h> + + apr { + compatible = "qcom,apr-v2"; + qcom,domain = <APR_DOMAIN_ADSP>; + #address-cells = <1>; + #size-cells = <0>; + + service@7 { + compatible = "qcom,q6asm"; + reg = <APR_SVC_ASM>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + dais { + compatible = "qcom,q6asm-dais"; + iommus = <&apps_smmu 0x1821 0x0>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@0 { + reg = <0>; + }; + + dai@1 { + reg = <1>; + }; + + dai@2 { + reg = <2>; + is-compress-dai; + direction = <1>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml new file mode 100644 index 000000000..fd567d204 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-clocks.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Qualcomm DSP LPASS Clock Controller binding + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP Clock Controller + +properties: + compatible: + enum: + - qcom,q6afe-clocks + - qcom,q6prm-lpass-clocks + + '#clock-cells': + const: 2 + description: + Clock Id is followed by clock coupling attributes. + 1 = for no coupled clock + 2 = for dividend of the coupled clock + 3 = for divisor of the coupled clock + 4 = for inverted and no couple clock + +required: + - compatible + - "#clock-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6afe.h> + apr { + compatible = "qcom,apr-v2"; + qcom,domain = <APR_DOMAIN_ADSP>; + #address-cells = <1>; + #size-cells = <0>; + + service@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + clock-controller { + compatible = "qcom,q6afe-clocks"; + #clock-cells = <2>; + }; + }; + }; + + - | + #include <dt-bindings/soc/qcom,gpr.h> + gpr { + compatible = "qcom,gpr"; + qcom,domain = <GPR_DOMAIN_ID_ADSP>; + #address-cells = <1>; + #size-cells = <0>; + + service@2 { + reg = <GPR_PRM_MODULE_IID>; + compatible = "qcom,q6prm"; + + clock-controller { + compatible = "qcom,q6prm-lpass-clocks"; + #clock-cells = <2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml new file mode 100644 index 000000000..e53fc0960 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml @@ -0,0 +1,203 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-ports.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Qualcomm DSP LPASS(Low Power Audio SubSystem) Audio Ports binding + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP LPASS Audio ports + +properties: + compatible: + enum: + - qcom,q6afe-dais + - qcom,q6apm-lpass-dais + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +#Digital Audio Interfaces +patternProperties: + '^dai@[0-9]+$': + 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/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6afe.h> + apr { + compatible = "qcom,apr-v2"; + #address-cells = <1>; + #size-cells = <0>; + qcom,domain = <APR_DOMAIN_ADSP>; + + service@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + 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>; + }; + }; + }; + }; + - | + #include <dt-bindings/soc/qcom,gpr.h> + gpr { + compatible = "qcom,gpr"; + #address-cells = <1>; + #size-cells = <0>; + qcom,domain = <GPR_DOMAIN_ID_ADSP>; + + service@1 { + compatible = "qcom,q6apm"; + reg = <GPR_APM_MODULE_IID>; + + dais { + compatible = "qcom,q6apm-lpass-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,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml new file mode 100644 index 000000000..70080d04d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -0,0 +1,314 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,sm8250.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. ASoC sound card drivers + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + This bindings describes Qualcomm SoC based sound cards + which uses LPASS internal codec for audio. + +properties: + compatible: + enum: + - lenovo,yoga-c630-sndcard + - qcom,apq8016-sbc-sndcard + - qcom,db845c-sndcard + - qcom,msm8916-qdsp6-sndcard + - qcom,qrb5165-rb5-sndcard + - qcom,sc8280xp-sndcard + - qcom,sdm845-sndcard + - qcom,sm8250-sndcard + - qcom,sm8450-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: 4 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - model + +allOf: + - if: + properties: + compatible: + contains: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + then: + properties: + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + required: + - compatible + - model + - reg + - reg-names + else: + properties: + reg: false + reg-names: false + +additionalProperties: false + +examples: + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound { + compatible = "qcom,qrb5165-rb5-sndcard"; + model = "Qualcomm-qrb5165-RB5-WSA8815-Speakers-DMIC0"; + audio-routing = "SpkrLeft IN", "WSA_SPK1 OUT", + "SpkrRight IN", "WSA_SPK2 OUT", + "VA DMIC0", "vdd-micb", + "VA DMIC1", "vdd-micb"; + + mm1-dai-link { + link-name = "MultiMedia0"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + mm2-dai-link { + link-name = "MultiMedia2"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA2>; + }; + }; + + mm3-dai-link { + link-name = "MultiMedia3"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA3>; + }; + }; + + hdmi-dai-link { + link-name = "HDMI Playback"; + cpu { + sound-dai = <&q6afedai TERTIARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + + codec { + sound-dai = <<9611_codec 0>; + }; + }; + + wsa-dai-link { + link-name = "WSA Playback"; + cpu { + sound-dai = <&q6afedai WSA_CODEC_DMA_RX_0>; + }; + + platform { + sound-dai = <&q6routing>; + }; + + codec { + sound-dai = <&left_spkr>, <&right_spkr>, <&swr0 0>, <&wsamacro>; + }; + }; + + va-dai-link { + link-name = "VA Capture"; + cpu { + sound-dai = <&q6afedai VA_CODEC_DMA_TX_0>; + }; + + platform { + sound-dai = <&q6routing>; + }; + + codec { + sound-dai = <&vamacro 0>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt new file mode 100644 index 000000000..1f75feec3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt @@ -0,0 +1,123 @@ +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 GPIO_ACTIVE_LOW>; + 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,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml new file mode 100644 index 000000000..8ca19f2b0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -0,0 +1,206 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd934x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WCD9340/WCD9341 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9340/WCD9341 Codec is a standalone Hi-Fi audio codec IC. + It has in-built Soundwire controller, pin controller, interrupt mux and + supports both I2S/I2C and SLIMbus audio interfaces. + +properties: + compatible: + const: slim217,250 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + reset-gpios: + description: GPIO spec for reset line to use + maxItems: 1 + + slim-ifc-dev: true + + clocks: + maxItems: 1 + + clock-names: + const: extclk + + vdd-buck-supply: + description: A reference to the 1.8V buck supply + + vdd-buck-sido-supply: + description: A reference to the 1.8V SIDO buck supply + + vdd-rx-supply: + description: A reference to the 1.8V rx supply + + vdd-tx-supply: + description: A reference to the 1.8V tx supply + + vdd-vbat-supply: + description: A reference to the vbat supply + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-micbias-supply: + description: A reference to the micbias supply + + qcom,micbias1-microvolt: + description: micbias1 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias2-microvolt: + description: micbias2 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias3-microvolt: + description: micbias3 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias4-microvolt: + description: micbias4 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,hphl-jack-type-normally-closed: + description: Indicates that HPHL jack switch type is normally closed + type: boolean + + qcom,ground-jack-type-normally-closed: + description: Indicates that Headset Ground switch type is normally closed + type: boolean + + qcom,mbhc-headset-vthreshold-microvolt: + description: Voltage threshold value for headset detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-headphone-vthreshold-microvolt: + description: Voltage threshold value for headphone detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-buttons-vthreshold-microvolt: + description: + Array of 8 Voltage threshold values corresponding to headset + button0 - button7 + minItems: 8 + maxItems: 8 + + clock-output-names: + const: mclk + + clock-frequency: + description: Clock frequency of output clk in Hz + + interrupt-controller: true + + '#interrupt-cells': + const: 1 + + '#clock-cells': + const: 0 + + '#sound-dai-cells': + const: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + gpio@42: + type: object + $ref: /schemas/gpio/qcom,wcd934x-gpio.yaml# + +patternProperties: + "^.*@[0-9a-f]+$": + type: object + description: | + WCD934x subnode for each slave devices. Bindings of each subnodes + depends on the specific driver providing the functionality and + documented in their respective bindings. + + properties: + reg: + maxItems: 1 + + required: + - reg + +required: + - compatible + - reg + - reset-gpios + - slim-ifc-dev + - interrupts + - interrupt-controller + - clock-frequency + - clock-output-names + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#interrupt-cells" + - "#clock-cells" + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + codec@1,0{ + compatible = "slim217,250"; + reg = <1 0>; + reset-gpios = <&tlmm 64 0>; + slim-ifc-dev = <&wcd9340_ifd>; + #sound-dai-cells = <1>; + interrupt-parent = <&tlmm>; + interrupts = <54 4>; + interrupt-controller; + #interrupt-cells = <1>; + #clock-cells = <0>; + clock-frequency = <9600000>; + clock-output-names = "mclk"; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headset-vthreshold-microvolt = <1700000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + clock-names = "extclk"; + clocks = <&rpmhcc 2>; + + #address-cells = <1>; + #size-cells = <1>; + + gpio@42 { + compatible = "qcom,wcd9340-gpio"; + reg = <0x42 0x2>; + gpio-controller; + #gpio-cells = <2>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml new file mode 100644 index 000000000..49a267b30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm SoundWire Slave devices on WCD9380/WCD9385 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. This bindings is for the + slave devices. + +properties: + compatible: + const: sdw20217010d00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between slave and master tx ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between slave and master rx ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml new file mode 100644 index 000000000..51547190f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -0,0 +1,153 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WCD9380/WCD9385 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. + +properties: + compatible: + enum: + - qcom,wcd9380-codec + - qcom,wcd9385-codec + + reset-gpios: + description: GPIO spec for reset line to use + maxItems: 1 + + us-euro-gpios: + description: GPIO spec for swapping gnd and mic segments + maxItems: 1 + + vdd-buck-supply: + description: A reference to the 1.8V buck supply + + vdd-rxtx-supply: + description: A reference to the 1.8V rx supply + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-mic-bias-supply: + description: A reference to the 3.8V mic bias supply + + qcom,tx-device: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A reference to Soundwire tx device phandle + + qcom,rx-device: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A reference to Soundwire rx device phandle + + qcom,micbias1-microvolt: + description: micbias1 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias2-microvolt: + description: micbias2 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias3-microvolt: + description: micbias3 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias4-microvolt: + description: micbias4 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,hphl-jack-type-normally-closed: + description: Indicates that HPHL jack switch type is normally closed + type: boolean + + qcom,ground-jack-type-normally-closed: + description: Indicates that Headset Ground switch type is normally closed + type: boolean + + qcom,mbhc-headset-vthreshold-microvolt: + description: Voltage threshold value for headset detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-headphone-vthreshold-microvolt: + description: Voltage threshold value for headphone detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-buttons-vthreshold-microvolt: + description: + Array of 8 Voltage threshold values corresponding to headset + button0 - button7 + minItems: 8 + maxItems: 8 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reset-gpios + - qcom,tx-device + - qcom,rx-device + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + codec { + compatible = "qcom,wcd9380-codec"; + reset-gpios = <&tlmm 32 0>; + #sound-dai-cells = <1>; + qcom,tx-device = <&wcd938x_tx>; + qcom,rx-device = <&wcd938x_rx>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + }; + + /* ... */ + + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml new file mode 100644 index 000000000..ea44d03e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa881x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WSA8810/WSA8815 Class-D Smart Speaker Amplifier + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + WSA8810 is a class-D smart speaker amplifier and WSA8815 + is a high-output power class-D smart speaker amplifier. + Their primary operating mode uses a SoundWire digital audio + interface. This binding is for SoundWire interface. + +properties: + compatible: + const: sdw10217201000 + + reg: + maxItems: 1 + + powerdown-gpios: + description: GPIO spec for Powerdown/Shutdown line to use + maxItems: 1 + + '#thermal-sensor-cells': + const: 0 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - powerdown-gpios + - "#thermal-sensor-cells" + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + soundwire@c2d0000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x0c2d0000 0x2000>; + + speaker@0,1 { + compatible = "sdw10217201000"; + reg = <0 1>; + powerdown-gpios = <&wcdpinctrl 2 0>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + + speaker@0,2 { + compatible = "sdw10217201000"; + reg = <0 2>; + powerdown-gpios = <&wcdpinctrl 2 0>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml new file mode 100644 index 000000000..6113f65f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa883x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for The Qualcomm WSA8830/WSA8832/WSA8835 + smart speaker amplifier + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + WSA883X is the Qualcomm Aqstic smart speaker amplifier + Their primary operating mode uses a SoundWire digital audio + interface. This binding is for SoundWire interface. + +properties: + compatible: + const: sdw10217020200 + + reg: + maxItems: 1 + + powerdown-gpios: + description: GPIO spec for Powerdown/Shutdown line to use + maxItems: 1 + + vdd-supply: + description: VDD Supply for the Codec + + '#thermal-sensor-cells': + const: 0 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - vdd-supply + - powerdown-gpios + - "#thermal-sensor-cells" + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + soundwire-controller@3250000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x3250000 0x2000>; + + speaker@0,1 { + compatible = "sdw10217020200"; + reg = <0 1>; + powerdown-gpios = <&tlmm 1 0>; + vdd-supply = <&vreg_s10b_1p8>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + + speaker@0,2 { + compatible = "sdw10217020200"; + reg = <0 2>; + powerdown-gpios = <&tlmm 89 0>; + vdd-supply = <&vreg_s10b_1p8>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml new file mode 100644 index 000000000..ea7d4900e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015p.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt1015p codec devicetree bindings + +maintainers: + - Tzung-Bi Shih <tzungbi@kernel.org> + +description: | + Rt1015p is a rt1015 variant which does not support I2C and + only supports S24, 48kHz, 64FS. + +properties: + compatible: + enum: + - realtek,rt1015p + - realtek,rt1019p + + sdb-gpios: + description: + GPIO used for shutdown control. + 0 means shut down; 1 means power on. + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + rt1015p: rt1015p { + compatible = "realtek,rt1015p"; + sdb-gpios = <&pio 175 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml new file mode 100644 index 000000000..ca5b8987b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -0,0 +1,121 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5682s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5682s codec devicetree bindings + +maintainers: + - Derek Fang <derek.fang@realtek.com> + +description: | + Rt5682s(ALC5682I-VS) is a rt5682i variant which supports I2C only. + +properties: + compatible: + const: realtek,rt5682s + + reg: + maxItems: 1 + description: I2C address of the device. + + interrupts: + maxItems: 1 + description: The CODEC's interrupt output. + + realtek,dmic1-data-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 data is not used + - 1 # using GPIO2 pin as dmic1 data pin + - 2 # using GPIO5 pin as dmic1 data pin + description: | + Specify which GPIO pin be used as DMIC1 data pin. + + realtek,dmic1-clk-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 clk is not used + - 1 # using GPIO1 pin as dmic1 clock pin + - 2 # using GPIO3 pin as dmic1 clock pin + description: | + Specify which GPIO pin be used as DMIC1 clk pin. + + realtek,jd-src: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # No JD is used + - 1 # using JD1 as JD source + description: | + Specify which JD source be used. + + realtek,ldo1-en-gpios: + description: | + The GPIO that controls the CODEC's LDO1_EN pin. + + realtek,dmic-clk-rate-hz: + description: | + Set the clock rate (hz) for the requirement of the particular DMIC. + + realtek,dmic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular DMIC. + + realtek,amic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular platform or AMIC. + + realtek,dmic-clk-driving-high: + type: boolean + description: | + Set the high driving of the DMIC clock out. + + clocks: + items: + - description: phandle and clock specifier for codec MCLK. + + clock-names: + items: + - const: mclk + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: Name given for DAI word clock and bit clock outputs. + +additionalProperties: false + +required: + - compatible + - reg + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,rt5682s"; + reg = <0x1a>; + interrupts = <6 IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio 2 GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml new file mode 100644 index 000000000..0dd3f7361 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,fsi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas FIFO-buffered Serial Interface (FSI) + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for FSI2 SoC + - items: + - enum: + - renesas,fsi2-sh73a0 # SH-Mobile AG5 + - renesas,fsi2-r8a7740 # R-Mobile A1 + - enum: + - renesas,sh_fsi2 + # for Generic + - items: + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + power-domains: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +patternProperties: + "^fsi(a|b),spdif-connection$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI is connected by S/PDIF + + "^fsi(a|b),stream-mode-support$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI supports 16bit stream mode + + "^fsi(a|b),use-internal-clock$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI uses internal clock when master mode + +required: + - compatible + - reg + - interrupts + - clocks + - power-domains + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/r8a7740-clock.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + sh_fsi2: sound@fe1f0000 { + compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2"; + reg = <0xfe1f0000 0x400>; + interrupts = <GIC_SPI 9 0x4>; + clocks = <&mstp3_clks R8A7740_CLK_FSI>; + power-domains = <&pd_a4mp>; + + #sound-dai-cells = <1>; + fsia,spdif-connection; + fsia,stream-mode-support; + fsia,use-internal-clock; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt new file mode 100644 index 000000000..b731f16ae --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -0,0 +1,255 @@ +Renesas R-Car sound + +============================================= +* Modules +============================================= + +Renesas R-Car and RZ/G sound is constructed from below modules +(for Gen2 or later) + + SCU : Sampling Rate Converter Unit + - SRC : Sampling Rate Converter + - CMD + - CTU : Channel Transfer Unit + - MIX : Mixer + - DVC : Digital Volume and Mute Function + SSIU : Serial Sound Interface Unit + SSI : Serial Sound Interface + +See detail of each module's channels, connection, limitation on datasheet + +============================================= +* Multi channel +============================================= + +Multi channel is supported by Multi-SSI, or TDM-SSI. + + Multi-SSI : 6ch case, you can use stereo x 3 SSI + TDM-SSI : 6ch case, you can use TDM + +============================================= +* Enable/Disable each modules +============================================= + +See datasheet to check SRC/CTU/MIX/DVC connect-limitation. +DT controls enabling/disabling module. +${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example. +This is example of + +Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] +Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] + +see "Example: simple sound card" + +You can use below. +${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. + + &src0 &ctu00 &mix0 &dvc0 &ssi0 + &src1 &ctu01 &mix1 &dvc1 &ssi1 + &src2 &ctu02 &ssi2 + &src3 &ctu03 &ssi3 + &src4 &ssi4 + &src5 &ctu10 &ssi5 + &src6 &ctu11 &ssi6 + &src7 &ctu12 &ssi7 + &src8 &ctu13 &ssi8 + &src9 &ssi9 + +============================================= +* SRC (Sampling Rate Converter) +============================================= + + [xx]Hz [yy]Hz + ------> [SRC] ------> + +SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes + + Asynchronous mode: input data / output data are based on different clocks. + you can use this mode on Playback / Capture + Synchronous mode: input data / output data are based on same clocks. + This mode will be used if system doesn't have its input clock, + for example digital TV case. + you can use this mode on Playback + +------------------ +** Asynchronous mode +------------------ + +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +see "Example: simple sound card for Asynchronous mode" + +------------------ +** Synchronous mode +------------------ + + > amixer set "SRC Out Rate" on + > aplay xxxx.wav + > amixer set "SRC Out Rate" 48000 + > amixer set "SRC Out Rate" 44100 + +============================================= +* CTU (Channel Transfer Unit) +============================================= + + [xx]ch [yy]ch + ------> [CTU] --------> + +CTU can convert [xx]ch to [yy]ch, or exchange outputed channel. +CTU conversion needs matrix settings. +For more detail information, see below + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - SCU Operation + - CMD Block + - Functional Blocks in CMD + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - Register Description + - CTUn Scale Value exx Register (CTUn_SVxxR) + + ${LINUX}/sound/soc/sh/rcar/ctu.c + - comment of header + +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +see "Example: simple sound card for channel convert" + +Ex) Exchange output channel + Input -> Output + 1ch -> 0ch + 0ch -> 1ch + + example of using matrix + output 0ch = (input 0ch x 0) + (input 1ch x 1) + output 1ch = (input 0ch x 1) + (input 1ch x 0) + + amixer set "CTU Reset" on + amixer set "CTU Pass" 9,10 + amixer set "CTU SV0" 0,4194304 + amixer set "CTU SV1" 4194304,0 + + example of changing connection + amixer set "CTU Reset" on + amixer set "CTU Pass" 2,1 + +============================================= +* MIX (Mixer) +============================================= + +MIX merges 2 sounds path. You can see 2 sound interface on system, +and these sounds will be merged by MIX. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 yyyy.wav + +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +Ex) + [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] + | + [MEM] -> [SRC2] -> [CTU03] -+ + +see "Example: simple sound card for MIXer" + +============================================= +* DVC (Digital Volume and Mute Function) +============================================= + +DVC controls Playback/Capture volume. + +Playback Volume + amixer set "DVC Out" 100% + +Capture Volume + amixer set "DVC In" 100% + +Playback Mute + amixer set "DVC Out Mute" on + +Capture Mute + amixer set "DVC In Mute" on + +Volume Ramp + amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + amixer set "DVC Out Ramp" on + aplay xxx.wav & + amixer set "DVC Out" 80% // Volume Down + amixer set "DVC Out" 100% // Volume Up + +============================================= +* SSIU (Serial Sound Interface Unit) +============================================= + +SSIU can avoid some under/over run error, because it has some buffer. +But you can't use it if SSI was PIO mode. +In DMA mode, you can select not to use SSIU by using "no-busif" via SSI. + +SSIU handles BUSIF which will be used for TDM Split mode. +This driver is assuming that audio-graph card will be used. + +TDM Split mode merges 4 sounds. You can see 4 sound interface on system, +and these sounds will be merged SSIU/SSI. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 xxxx.wav & + aplay -D plughw:0,2 xxxx.wav & + aplay -D plughw:0,3 xxxx.wav + + 2ch 8ch + [MEM] -> [SSIU 30] -+-> [SSIU 3] --> [Codec] + 2ch | + [MEM] -> [SSIU 31] -+ + 2ch | + [MEM] -> [SSIU 32] -+ + 2ch | + [MEM] -> [SSIU 33] -+ + +see "Example: simple sound card for TDM Split" + +============================================= +* SSI (Serial Sound Interface) +============================================= + +** PIO mode + +You can use PIO mode which is for connection check by using. +Note: The system will drop non-SSI modules in PIO mode +even though if DT is selecting other modules. + + &ssi0 { + pio-transfer + }; + +** DMA mode without SSIU + +You can use DMA without SSIU. +Note: under/over run, or noise are likely to occur + + &ssi0 { + no-busif; + }; + +** PIN sharing + +Each SSI can share WS pin. It is based on platform. +This is example if SSI1 want to share WS pin with SSI0 + + &ssi1 { + shared-pin; + }; + +** Multi-SSI + +You can use Multi-SSI. +This is example of SSI0/SSI1/SSI2 (= for 6ch) + +see "Example: simple sound card for Multi channel" + +** TDM-SSI + +You can use TDM with SSI. +This is example of TDM 6ch. +Driver can automatically switches TDM <-> stereo mode in this case. + +see "Example: simple sound card for TDM" diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 000000000..679a246dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,463 @@ +# 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 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7742 # RZ/G1H + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a774e1 # RZ/G2H + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77961 # R-Car M3-W+ + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 1 + maxItems: 5 + + reg-names: + minItems: 1 + maxItems: 5 + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + minItems: 1 + maxItems: 11 + + reset-names: + minItems: 1 + maxItems: 11 + + clocks: + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + minItems: 1 + maxItems: 31 + + clock-names: + description: List of necessary clock names. + minItems: 1 + maxItems: 31 + items: + oneOf: + - const: ssi-all + - pattern: '^ssi\.[0-9]$' + - pattern: '^src\.[0-9]$' + - pattern: '^mix\.[0-1]$' + - pattern: '^ctu\.[0-1]$' + - pattern: '^dvc\.[0-1]$' + - pattern: '^clk_(a|b|c|i)$' + + ports: + $ref: /schemas/graph.yaml#/properties/ports + patternProperties: + port(@[0-9a-f]+)?: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + rcar_sound,dvc: + description: DVC subnode. + type: object + patternProperties: + "^dvc-[0-1]$": + type: object + 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]$": + type: object + additionalProperties: false + + properties: + interrupts: + maxItems: 1 + dmas: + maxItems: 2 + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + rcar_sound,ssiu: + description: SSIU subnode. + type: object + patternProperties: + "^ssiu-[0-9]+$": + 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 + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + rcar_sound,dai: + description: DAI subnode. + type: object + patternProperties: + "^dai([0-9]+)?$": + 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 + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +allOf: + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen1 + then: + properties: + reg: + maxItems: 3 + reg-names: + maxItems: 3 + items: + enum: + - scu + - ssi + - adg + else: + properties: + reg: + maxItems: 5 + reg-names: + maxItems: 5 + items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0xec500000 0x1000>, /* SCU */ + <0xec5a0000 0x100>, /* ADG */ + <0xec540000 0x1000>, /* SSIU */ + <0xec541000 0x1280>, /* SSI */ + <0xec740000 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* MIX1, MIX0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CTU1, CTU0 */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>, /* DVC0, DVC1 */ + <&audio_clk_a>, <&audio_clk_b>, /* CLKA, CLKB */ + <&audio_clk_c>, <&audio_clk_i>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "mix.1", "mix.0", + "ctu.1", "ctu.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5>, <&src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* assume audio-graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0>, <&src0>, <&dvc0>; + capture = <&ssi1>, <&src1>, <&dvc1>; + }; + }; + }; + + + /* assume audio-graph */ + codec { + port { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml new file mode 100644 index 000000000..0d9840375 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.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/renesas,rz-ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2) + +maintainers: + - Biju Das <biju.das.jz@bp.renesas.com> + +properties: + compatible: + items: + - enum: + - renesas,r9a07g043-ssi # RZ/G2UL + - renesas,r9a07g044-ssi # RZ/G2{L,LC} + - renesas,r9a07g054-ssi # RZ/V2L + - const: renesas,rz-ssi + + reg: + maxItems: 1 + + interrupts: + maxItems: 4 + + interrupt-names: + items: + - const: int_req + - const: dma_rx + - const: dma_tx + - const: dma_rt + + clocks: + maxItems: 4 + + clock-names: + items: + - const: ssi + - const: ssi_sfr + - const: audio_clk1 + - const: audio_clk2 + + power-domains: + maxItems: 1 + + resets: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + description: + The first cell represents a phandle to dmac. + The second cell specifies the encoded MID/RID values of the SSI port + connected to the DMA client and the slave channel configuration + parameters. + bits[0:9] - Specifies MID/RID value of a SSI channel as below + MID/RID value of SSI rx0 = 0x256 + MID/RID value of SSI tx0 = 0x255 + MID/RID value of SSI rx1 = 0x25a + MID/RID value of SSI tx1 = 0x259 + MID/RID value of SSI rt2 = 0x25f + MID/RID value of SSI rx3 = 0x262 + MID/RID value of SSI tx3 = 0x261 + bit[10] - HIEN = 1, Detects a request in response to the rising edge + of the signal + bit[11] - LVL = 0, Detects based on the edge + bits[12:14] - AM = 2, Bus cycle mode + bit[15] - TM = 0, Single transfer mode + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: rt + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - interrupts + - interrupt-names + - clocks + - clock-names + - resets + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/r9a07g044-cpg.h> + + ssi0: ssi@10049c00 { + compatible = "renesas,r9a07g044-ssi", + "renesas,rz-ssi"; + reg = <0x10049c00 0x400>; + interrupts = <GIC_SPI 326 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 327 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 328 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 329 IRQ_TYPE_EDGE_RISING>; + interrupt-names = "int_req", "dma_rx", "dma_tx", "dma_rt"; + clocks = <&cpg CPG_MOD R9A07G044_SSI0_PCLK2>, + <&cpg CPG_MOD R9A07G044_SSI0_PCLK_SFR>, + <&audio_clk1>, + <&audio_clk2>; + clock-names = "ssi", "ssi_sfr", "audio_clk1", "audio_clk2"; + power-domains = <&cpg>; + resets = <&cpg R9A07G044_SSI0_RST_M2_REG>; + dmas = <&dmac 0x2655>, + <&dmac 0x2656>; + dma-names = "tx", "rx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml new file mode 100644 index 000000000..5655ca568 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/richtek,rt9120.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Richtek RT9120 Class-D audio amplifier + +maintainers: + - ChiYuan Huang <cy_huang@richtek.com> + +description: | + The RT9120 is a high efficiency, I2S-input, stereo audio power amplifier + delivering 2*20W into 8 Ohm BTL speaker loads. It supports the wide input + voltage range from 4.5V to 26.4V to meet the need on most common + applications like as TV, monitors. home entertainment, electronic music + equipment. + +properties: + compatible: + enum: + - richtek,rt9120 + + reg: + description: I2C device address + maxItems: 1 + + pwdnn-gpios: + description: GPIO used for power down, low active + maxItems: 1 + + dvdd-supply: + description: | + Supply for the default on DVDD power, voltage domain must be 3P3V or 1P8V + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - dvdd-supply + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt9120@1a { + compatible = "richtek,rt9120"; + reg = <0x1a>; + pwdnn-gpios = <&gpio26 2 0>; + dvdd-supply = <&vdd_io_reg>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml new file mode 100644 index 000000000..6a7c004be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,i2s-tdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S/TDM Controller + +description: + The Rockchip I2S/TDM Controller is a Time Division Multiplexed + audio interface found in various Rockchip SoCs, allowing up + to 8 channels of audio over a serial interface. + +maintainers: + - Nicolas Frattaroli <frattaroli.nicolas@gmail.com> + +properties: + compatible: + enum: + - rockchip,px30-i2s-tdm + - rockchip,rk1808-i2s-tdm + - rockchip,rk3308-i2s-tdm + - rockchip,rk3568-i2s-tdm + - rockchip,rv1126-i2s-tdm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + maxItems: 2 + items: + enum: + - rx + - tx + + clocks: + minItems: 3 + items: + - description: clock for TX + - description: clock for RX + - description: AHB clock driving the interface + - description: + Parent clock for mclk_tx (only required when using mclk-calibrate) + - description: + Parent clock for mclk_rx (only required when using mclk-calibrate) + - description: + Clock for sample rates that are an integer multiple of 8000 + (only required when using mclk-calibrate) + - description: + Clock for sample rates that are an integer multiple of 11025 + (only required when using mclk-calibrate) + + clock-names: + minItems: 3 + items: + - const: mclk_tx + - const: mclk_rx + - const: hclk + - const: mclk_tx_src + - const: mclk_rx_src + - const: mclk_root0 + - const: mclk_root1 + + resets: + minItems: 1 + maxItems: 2 + description: resets for the tx and rx directions + + reset-names: + minItems: 1 + maxItems: 2 + items: + enum: + - tx-m + - rx-m + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + rockchip,trcm-sync-tx-only: + type: boolean + description: Use TX BCLK/LRCK for both TX and RX. + + rockchip,trcm-sync-rx-only: + type: boolean + description: Use RX BCLK/LRCK for both TX and RX. + + "#sound-dai-cells": + const: 0 + + rockchip,i2s-rx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of I2S RX sdis to I2S data bus lines. + By default, they are mapped one-to-one. + rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0. + maxItems: 4 + items: + enum: [0, 1, 2, 3] + + rockchip,i2s-tx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of I2S TX sdos to I2S data bus lines. + By default, they are mapped one-to-one. + rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0. + maxItems: 4 + items: + enum: [0, 1, 2, 3] + + rockchip,io-multiplex: + description: + Specify that the GPIO lines on the I2S bus are multiplexed such that + the direction (input/output) needs to be dynamically adjusted. + type: boolean + + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + - resets + - reset-names + - rockchip,grf + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3568-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + i2s@fe410000 { + compatible = "rockchip,rk3568-i2s-tdm"; + reg = <0x0 0xfe410000 0x0 0x1000>; + interrupts = <GIC_SPI 53 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru MCLK_I2S1_8CH_TX>, <&cru MCLK_I2S1_8CH_RX>, + <&cru HCLK_I2S1_8CH>; + clock-names = "mclk_tx", "mclk_rx", "hclk"; + dmas = <&dmac1 3>, <&dmac1 2>; + dma-names = "rx", "tx"; + resets = <&cru SRST_M_I2S1_8CH_TX>, <&cru SRST_M_I2S1_8CH_RX>; + reset-names = "tx-m", "rx-m"; + rockchip,trcm-sync-tx-only; + rockchip,grf = <&grf>; + #sound-dai-cells = <0>; + pinctrl-names = "default"; + pinctrl-0 = + <&i2s1m0_sclktx + &i2s1m0_sclkrx + &i2s1m0_lrcktx + &i2s1m0_lrckrx + &i2s1m0_sdi0 + &i2s1m0_sdi1 + &i2s1m0_sdi2 + &i2s1m0_sdi3 + &i2s1m0_sdo0 + &i2s1m0_sdo1 + &i2s1m0_sdo2 + &i2s1m0_sdo3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml new file mode 100644 index 000000000..22e1cf6c0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,pdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip PDM controller + +description: + The Pulse Density Modulation Interface Controller (PDMC) is + a PDM interface controller and decoder that support PDM format. + It integrates a clock generator driving the PDM microphone + and embeds filters which decimate the incoming bit stream to + obtain most common audio rates. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + enum: + - rockchip,pdm + - rockchip,px30-pdm + - rockchip,rk1808-pdm + - rockchip,rk3308-pdm + - rockchip,rk3568-pdm + - rockchip,rv1126-pdm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for PDM controller + - description: clock for PDM BUS + + clock-names: + items: + - const: pdm_clk + - const: pdm_hclk + + dmas: + maxItems: 1 + + dma-names: + items: + - const: rx + + power-domains: + maxItems: 1 + + resets: + items: + - description: reset for PDM controller + + reset-names: + items: + - const: pdm-m + + rockchip,path-map: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of PDM SDIx to PDM PATHx. + By default, they are mapped one-to-one. + maxItems: 4 + uniqueItems: true + items: + enum: [ 0, 1, 2, 3 ] + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3328-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + + pdm@ff040000 { + compatible = "rockchip,pdm"; + reg = <0x0 0xff040000 0x0 0x1000>; + interrupts = <GIC_SPI 82 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_PDM>, <&cru HCLK_PDM>; + clock-names = "pdm_clk", "pdm_hclk"; + dmas = <&dmac 16>; + dma-names = "rx"; + #sound-dai-cells = <0>; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pdmm0_clk + &pdmm0_sdi0 + &pdmm0_sdi1 + &pdmm0_sdi2 + &pdmm0_sdi3>; + pinctrl-1 = <&pdmm0_clk_sleep + &pdmm0_sdi0_sleep + &pdmm0_sdi1_sleep + &pdmm0_sdi2_sleep + &pdmm0_sdi3_sleep>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt new file mode 100644 index 000000000..e5430d1d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt @@ -0,0 +1,36 @@ +ROCKCHIP RK3288 with HDMI and analog audio + +Required properties: +- compatible: "rockchip,rk3288-hdmi-analog" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the analog audio codec. +- rockchip,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. For this driver the first string should always be + "Analog". + +Optionnal 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 + (when the analog output is an headphone). +- pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: + +sound { + compatible = "rockchip,rk3288-hdmi-analog"; + rockchip,model = "Analog audio output"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&es8388>; + rockchip,routing = "Analog", "LOUT2", + "Analog", "ROUT2"; + rockchip,hp-en-gpios = <&gpio8 0 GPIO_ACTIVE_HIGH>; + rockchip,hp-det-gpios = <&gpio7 7 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphone>; +}; + 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 000000000..75b3b33b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 000000000..72d3cf4c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,22 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the audio codecs + +Optional properties: +- dmic-wakeup-delay-ms : specify delay time (ms) for DMIC ready. + If this option is specified, which means it's required dmic need + delay for DMIC to ready so that rt5514 can avoid recording before + DMIC send valid data + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; + dmic-wakeup-delay-ms = <20>; +}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000..7e36e389e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,132 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk1808-i2s + - rockchip,rk3036-i2s + - rockchip,rk3128-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3308-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - rockchip,rv1126-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + oneOf: + - const: rx + - items: + - const: tx + - const: rx + + 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 + + rockchip,capture-channels: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3288-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98090.txt b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt new file mode 100644 index 000000000..e9c58b204 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt @@ -0,0 +1,42 @@ +ROCKCHIP with MAX98090 CODEC + +Required properties: +- compatible: "rockchip,rockchip-audio-max98090" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC + +Optional properties: +- rockchip,audio-codec: The phandle of the MAX98090 audio codec. +- rockchip,headset-codec: The phandle of Ext chip for jack detection. This is + required if there is rockchip,audio-codec. +- rockchip,hdmi-codec: The phandle of HDMI device for HDMI codec. + +Example: + +/* For max98090-only board. */ +sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&max98090>; + rockchip,headset-codec = <&headsetcodec>; +}; + +/* For HDMI-only board. */ +sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,hdmi-codec = <&hdmi>; +}; + +/* For max98090 plus HDMI board. */ +sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&max98090>; + rockchip,headset-codec = <&headsetcodec>; + rockchip,hdmi-codec = <&hdmi>; +}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt b/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt new file mode 100644 index 000000000..411a62b3f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt @@ -0,0 +1,17 @@ +ROCKCHIP with RT5645/RT5650 CODECS + +Required properties: +- compatible: "rockchip,rockchip-audio-rt5645" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the RT5645/RT5650 audio codec + +Example: + +sound { + compatible = "rockchip,rockchip-audio-rt5645"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&rt5645>; +}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..d0a24bf92 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - const: rockchip,rk3568-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - rockchip,rk3308-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + power-domains: + maxItems: 1 + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000..859ce64da --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro@katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt new file mode 100644 index 000000000..02d53b9aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1011.txt @@ -0,0 +1,42 @@ +RT1011 Mono Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1011". + +- reg : The I2C address of the device. This I2C address decide by + two input pins (ASEL1 and ASEL2). + ------------------------------------- + | ASEL2 | ASEL1 | Address | + ------------------------------------- + | 0 | 0 | 0x38 | + ------------------------------------- + | 0 | 1 | 0x39 | + ------------------------------------- + | 1 | 0 | 0x3a | + ------------------------------------- + | 1 | 1 | 0x3b | + ------------------------------------- + +Optional properties: + +- realtek,temperature_calib + u32. The temperature was measured while doing the calibration. Units: Celsius degree + +- realtek,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + +Pins on the device (for linking into audio routes) for RT1011: + + * SPO + +Example: + +rt1011: codec@38 { + compatible = "realtek,rt1011"; + reg = <0x38>; + realtek,temperature_calib = <25>; + realtek,r0_calib = <0x224050>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt new file mode 100644 index 000000000..e498966d4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -0,0 +1,23 @@ +RT1015 Mono Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1015". + +- reg : The I2C address of the device. + +Optional properties: + +- realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. + +Example: + +rt1015: codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt1016.txt b/Documentation/devicetree/bindings/sound/rt1016.txt new file mode 100644 index 000000000..2310f8ff2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1016.txt @@ -0,0 +1,17 @@ +RT1016 Stereo Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1016". + +- reg : The I2C address of the device. + + +Example: + +rt1016: codec@1a { + compatible = "realtek,rt1016"; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt1019.yaml b/Documentation/devicetree/bindings/sound/rt1019.yaml new file mode 100644 index 000000000..3d5a91a94 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1019.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rt1019.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1019 Mono Class-D Audio Amplifier + +maintainers: + - jack.yu@realtek.com + +properties: + compatible: + const: realtek,rt1019 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt1019: codec@28 { + compatible = "realtek,rt1019"; + reg = <0x28>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1308.txt b/Documentation/devicetree/bindings/sound/rt1308.txt new file mode 100644 index 000000000..2d46084af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1308.txt @@ -0,0 +1,17 @@ +RT1308 audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1308". + +- reg : The I2C address of the device. + + +Example: + +rt1308: rt1308@10 { + compatible = "realtek,rt1308"; + reg = <0x10>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt new file mode 100644 index 000000000..791a1bd76 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -0,0 +1,33 @@ +RT274 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt274". + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + + +Pins on the device (for linking into audio routes) for RT274: + + * DMIC1 Pin + * DMIC2 Pin + * MIC + * LINE1 + * LINE2 + * HPO Pin + * SPDIF + * LINE3 + +Example: + +rt274: codec@1c { + compatible = "realtek,rt274"; + reg = <0x1c>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt new file mode 100644 index 000000000..d2cc171f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -0,0 +1,37 @@ +RT5514 audio CODEC + +This device supports both I2C and SPI. + +Required properties: + +- compatible : "realtek,rt5514". + +- reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + +- realtek,dmic-init-delay-ms + Set the DMIC initial delay (ms) to wait it ready for I2C. + +Pins on the device (for linking into audio routes) for I2C: + + * DMIC1L + * DMIC1R + * DMIC2L + * DMIC2R + * AMICL + * AMICR + +Example: + +rt5514: codec@57 { + compatible = "realtek,rt5514"; + reg = <0x57>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt new file mode 100644 index 000000000..540a4bf25 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -0,0 +1,32 @@ +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/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt new file mode 100644 index 000000000..56bc85232 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5631.txt @@ -0,0 +1,48 @@ +ALC5631/RT5631 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "realtek,alc5631" or "realtek,rt5631" + + - reg : the I2C address of the device. + +Pins on the device (for linking into audio routes): + + * SPK_OUT_R_P + * SPK_OUT_R_N + * SPK_OUT_L_P + * SPK_OUT_L_N + * HP_OUT_L + * HP_OUT_R + * AUX_OUT2_LP + * AUX_OUT2_RN + * AUX_OUT1_LP + * AUX_OUT1_RN + * AUX_IN_L_JD + * AUX_IN_R_JD + * MONO_IN_P + * MONO_IN_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MONO_OUT_P + * MONO_OUT_N + * MICBIAS1 + * MICBIAS2 + +Example: + +alc5631: audio-codec@1a { + compatible = "realtek,alc5631"; + reg = <0x1a>; +}; + +or + +rt5631: audio-codec@1a { + compatible = "realtek,rt5631"; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt new file mode 100644 index 000000000..ff1228713 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -0,0 +1,94 @@ +RT5640/RT5639 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5640" or "realtek,rt5639". + +- 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" + +- realtek,in1-differential +- realtek,in2-differential +- realtek,in3-differential + Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN1P pin as dmic1 data pin + 2: using GPIO3 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN1N pin as dmic2 data pin + 2: using GPIO4 pin as dmic2 data pin + +- realtek,jack-detect-source + u32. Valid values: + 0: jack-detect is not used + 1: Use GPIO1 for jack-detect + 2: Use JD1_IN4P for jack-detect + 3: Use JD2_IN4N for jack-detect + 4: Use GPIO2 for jack-detect + 5: Use GPIO3 for jack-detect + 6: Use GPIO4 for jack-detect + +- realtek,jack-detect-not-inverted + bool. Normal jack-detect switches give an inverted signal, set this bool + in the rare case you've a jack-detect switch which is not inverted. + +- realtek,over-current-threshold-microamp + u32, micbias over-current detection threshold in µA, valid values are + 600, 1500 and 2000µA. + +- realtek,over-current-scale-factor + u32, micbias over-current detection scale-factor, valid values are: + 0: Scale current by 0.5 + 1: Scale current by 0.75 + 2: Scale current by 1.0 + 3: Scale current by 1.5 + +Pins on the device (for linking into audio routes) for RT5639/RT5640: + + * DMIC1 + * DMIC2 + * MICBIAS1 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * HPOL + * HPOR + * LOUTL + * LOUTR + * SPOLP + * SPOLN + * SPORP + * SPORN + +Additional pins on the device for RT5640: + + * MONOP + * MONON + +Example: + +rt5640 { + compatible = "realtek,rt5640"; + reg = <0x1c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5645.txt b/Documentation/devicetree/bindings/sound/rt5645.txt new file mode 100644 index 000000000..41a62fd2a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5645.txt @@ -0,0 +1,76 @@ +RT5650/RT5645 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5645" or "realtek,rt5650". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +- avdd-supply: Power supply for AVDD, providing 1.8V. + +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + +Optional properties: + +- hp-detect-gpios: + a GPIO spec for the external headphone detect pin. If jd-mode = 0, + we will get the JD status by getting the value of hp-detect-gpios. + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN2P pin as dmic1 data pin + 2: using GPIO6 pin as dmic1 data pin + 3: using GPIO10 pin as dmic1 data pin + 4: using GPIO12 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN2N pin as dmic2 data pin + 2: using GPIO5 pin as dmic2 data pin + 3: using GPIO11 pin as dmic2 data pin + +-- realtek,jd-mode : The JD mode of rt5645/rt5650 + 0 : rt5645/rt5650 JD function is not used + 1 : Mode-0 (VDD=3.3V), two port jack detection + 2 : Mode-1 (VDD=3.3V), one port jack detection + 3 : Mode-2 (VDD=1.8V), one port jack detection + +Pins on the device (for linking into audio routes) for RT5645/RT5650: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * Haptic Generator + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDM1L + * PDM1R + * SPOL + * SPOR + +Example: + +codec: rt5650@1a { + compatible = "realtek,rt5650"; + reg = <0x1a>; + hp-detect-gpios = <&gpio 19 0>; + interrupt-parent = <&gpio>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + realtek,dmic-en = "true"; + realtek,en-jd-func = "true"; + realtek,jd-mode = <3>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt new file mode 100644 index 000000000..56e736a1c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5651.txt @@ -0,0 +1,63 @@ +RT5651 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5651". + +- reg : The I2C address of the device. + +Optional properties: + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic-en + Boolean. true if dmic is used. + +- realtek,jack-detect-source + u32. Valid values: + 1: Use JD1_1 pin for jack-detect + 2: Use JD1_2 pin for jack-detect + 3: Use JD2 pin for jack-detect + +- realtek,jack-detect-not-inverted + bool. Normal jack-detect switches give an inverted (active-low) signal, + set this bool in the rare case you've a jack-detect switch which is not + inverted. + +- realtek,over-current-threshold-microamp + u32, micbias over-current detection threshold in µA, valid values are + 600, 1500 and 2000µA. + +- realtek,over-current-scale-factor + u32, micbias over-current detection scale-factor, valid values are: + 0: Scale current by 0.5 + 1: Scale current by 0.75 + 2: Scale current by 1.0 + 3: Scale current by 1.5 + +Pins on the device (for linking into audio routes) for RT5651: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN2P + * IN2N + * IN3P + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDML + * PDMR + +Example: + +rt5651: codec@1a { + compatible = "realtek,rt5651"; + reg = <0x1a>; + realtek,dmic-en = "true"; + realtek,in2-diff = "false"; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt new file mode 100644 index 000000000..013f534fa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -0,0 +1,89 @@ +RT5659/RT5658 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5659" or "realtek,rt5658". + +- 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" + +- realtek,in1-differential +- realtek,in3-differential +- realtek,in4-differential + Boolean. Indicate MIC1/3/4 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN2N pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + 3: using GPIO9 pin as dmic1 data pin + 4: using GPIO11 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN2P pin as dmic2 data pin + 2: using GPIO6 pin as dmic2 data pin + 3: using GPIO10 pin as dmic2 data pin + 4: using GPIO12 pin as dmic2 data pin + +- realtek,jd-src + 0: No JD is used + 1: using JD3 as JD source + 2: JD source for Intel HDA header + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,reset-gpios : The GPIO that controls the CODEC's RESET pin. + +- sound-name-prefix: Please refer to name-prefix.yaml + +- ports: A Codec may have a single or multiple I2S interfaces. These + interfaces on Codec side can be described under 'ports' or 'port'. + When the SoC or host device is connected to multiple interfaces of + the Codec, the connectivity can be described using 'ports' property. + If a single interface is used, then 'port' can be used. The usage + depends on the platform or board design. + Please refer to Documentation/devicetree/bindings/graph.txt + +Pins on the device (for linking into audio routes) for RT5659/RT5658: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * IN4P + * IN4N + * HPOL + * HPOR + * SPOL + * SPOR + * LOUTL + * LOUTR + * MONOOUT + * PDML + * PDMR + * SPDIF + +Example: + +rt5659 { + compatible = "realtek,rt5659"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + 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/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt new file mode 100644 index 000000000..30be5f921 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5660.txt @@ -0,0 +1,47 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt new file mode 100644 index 000000000..2a55e9133 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -0,0 +1,60 @@ +RT5663 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5663". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +- avdd-supply: Power supply for AVDD, providing 1.8V. + +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + +Optional properties: + +- "realtek,dc_offset_l_manual" +- "realtek,dc_offset_r_manual" +- "realtek,dc_offset_l_manual_mic" +- "realtek,dc_offset_r_manual_mic" + Based on the different PCB layout, add the manual offset value to + compensate the DC offset for each L and R channel, and they are different + between headphone and headset. +- "realtek,impedance_sensing_num" + The matrix row number of the impedance sensing table. + 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 + R channel accordingly. Example is shown as following. + < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0 + 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0> + The first and second column are defined for the impedance range. If the + detected impedance value is in the range, then the volume value of the + third column will be set to codec. In our codec design, each volume value + should compensate different DC offset to avoid the pop sound, and it is + also different between headphone and headset. In the example, the + "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of + impedance in the impedance sensing function. + +Pins on the device (for linking into audio routes) for RT5663: + + * IN1P + * IN1N + * IN2P + * IN2N + * HPOL + * HPOR + +Example: + +rt5663: codec@12 { + compatible = "realtek,rt5663"; + reg = <0x12>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt new file mode 100644 index 000000000..f6ca96b4c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -0,0 +1,68 @@ +RT5665/RT5666 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5665", "realtek,rt5666". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential +- realtek,in3-differential +- realtek,in4-differential + Boolean. Indicate MIC1/2/3/4 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO4 pin as dmic1 data pin + 2: using IN2N pin as dmic2 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using GPIO5 pin as dmic2 data pin + 2: using IN2P pin as dmic2 data pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5659/RT5658: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * IN4P + * IN4N + * HPOL + * HPOR + * LOUTL + * LOUTR + * MONOOUT + * PDML + * PDMR + +Example: + +rt5659 { + compatible = "realtek,rt5665"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + 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 new file mode 100644 index 000000000..a2b7e9a2f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -0,0 +1,50 @@ +RT5668B audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5668b" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5668B: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5668 { + compatible = "realtek,rt5668b"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt new file mode 100644 index 000000000..da2430099 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -0,0 +1,78 @@ +RT5677 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5677". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +- 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: + +- realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low. + +- realtek,in1-differential +- realtek,in2-differential +- realtek,lout1-differential +- realtek,lout2-differential +- realtek,lout3-differential + Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential, + rather than single-ended. + +- realtek,gpio-config + Array of six 8bit elements that configures GPIO. + 0 - floating (reset value) + 1 - pull down + 2 - pull up + +- realtek,jd1-gpio + Configures GPIO Mic Jack detection 1. + Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively. + +- realtek,jd2-gpio +- realtek,jd3-gpio + Configures GPIO Mic Jack detection 2 and 3. + Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively. + +Pins on the device (for linking into audio routes): + + * IN1P + * IN1N + * IN2P + * IN2N + * MICBIAS1 + * DMIC1 + * DMIC2 + * DMIC3 + * DMIC4 + * LOUT1 + * LOUT2 + * LOUT3 + +Example: + +rt5677 { + compatible = "realtek,rt5677"; + reg = <0x2c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; + + gpio-controller; + #gpio-cells = <2>; + + realtek,pow-ldo2-gpio = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; + realtek,reset-gpio = <&gpio TEGRA_GPIO(BB, 3) GPIO_ACTIVE_LOW>; + realtek,in1-differential = "true"; + realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ + realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ +}; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt new file mode 100644 index 000000000..6b87db683 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -0,0 +1,78 @@ +RT5682 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5682" or "realtek,rt5682i" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +- realtek,btndet-delay + The debounce delay for push button. + The delay time is realtek,btndet-delay value multiple of 8.192 ms. + If absent, the default is 16. + +- #clock-cells : Should be set to '<1>', wclk and bclk sources provided. +- clock-output-names : Name given for DAI clocks output. + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- realtek,dmic-clk-rate-hz : Set the clock rate (hz) for the requirement of + the particular DMIC. + +- realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of + the particular DMIC. + +- realtek,dmic-clk-driving-high : Set the high driving of the DMIC clock out. + +- #sound-dai-cells: Should be set to '<1>'. + +Pins on the device (for linking into audio routes) for RT5682: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5682 { + compatible = "realtek,rt5682i"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; + realtek,btndet-delay = <16>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; +}; diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml new file mode 100644 index 000000000..447e013f6 --- /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> + +properties: + compatible: + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + 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 + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the WM8994's pins (as + documented in its binding), and the jacks on the board - + For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic, + or FM In + For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic, + or HeadsetMic + + extcon: + description: Extcon phandle for dock detection + + main-micbias-supply: + description: Supply for the micbias on the main mic + + headset-micbias-supply: + description: Supply for the micbias on the headset mic + + earpath-sel-gpios: + maxItems: 1 + description: GPIO for switching between tv-out and mic paths + + headset-detect-gpios: + maxItems: 1 + description: GPIO for detection of headset insertion + + headset-key-gpios: + maxItems: 1 + description: GPIO for detection of headset key press + + io-channels: + maxItems: 1 + description: IO channel to read micbias voltage for headset detection + + io-channel-names: + const: headset-detect + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - extcon + - main-micbias-supply + - headset-micbias-supply + - earpath-sel-gpios + - headset-detect-gpios + - headset-key-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,fascinate4g-wm8994"; + + model = "Fascinate4G"; + + extcon = <&fsa9480>; + + main-micbias-supply = <&main_micbias_reg>; + headset-micbias-supply = <&headset_micbias_reg>; + + earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>; + + io-channels = <&adc 3>; + io-channel-names = "headset-detect"; + headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>; + headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "LINE", "LINEOUT2N", + "LINE", "LINEOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + + "IN1RP", "Headset Mic", + "IN1RN", "Headset Mic"; + + pinctrl-names = "default"; + pinctrl-0 = <&headset_det &earpath_sel>; + + cpu { + sound-dai = <&i2s0>, <&bt_codec>; + }; + + codec { + sound-dai = <&wm8994>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,arndale.yaml b/Documentation/devicetree/bindings/sound/samsung,arndale.yaml new file mode 100644 index 000000000..9bc4585bb --- /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. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - samsung,audio-codec + - samsung,audio-cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,arndale-rt5631"; + samsung,audio-cpu = <&i2s0>; + samsung,audio-codec = <&rt5631>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 000000000..31095913e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,112 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + 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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + maxItems: 1 + description: GPIO pin for FM selection + + lineout-sel-gpios: + maxItems: 1 + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000..7b4e08dde --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + clocks: true + + cpu: + type: object + properties: + sound-dai: + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + items: + - description: phandle of the MAX98090 CODEC + - description: phandle of the HDMI IP block node + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + +required: + - compatible + - model + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml new file mode 100644 index 000000000..ac151d3c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + $ref: /schemas/types.yaml#/definitions/phandle + +required: + - compatible + - samsung,audio-codec + - samsung,i2s-controller + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,smdk-wm8994"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&wm8994>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml new file mode 100644 index 000000000..3d49aa4c9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - codec + - cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&max98095 0>, <&hdmi>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml new file mode 100644 index 000000000..491e08019 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + audio-amplifier: + description: Phandle to the MAX98504 amplifier. + $ref: /schemas/types.yaml#/definitions/phandle + + audio-codec: + description: Phandles to the codecs. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - description: Phandle to the WM5110 audio codec. + - description: Phandle to the HDMI transmitter node. + + samsung,audio-routing: + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are the + WM5110's and MAX98504's pins and the jacks on the board: HP, SPK, Main + Mic, Sub Mic, Third Mic, Headset Mic. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + i2s-controller: + description: Phandles to the I2S controllers. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - description: Phandle to I2S0. + - description: Phandle to I2S1. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + + model: + description: The user-visible name of this sound complex. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - audio-amplifier + - audio-codec + - samsung,audio-routing + - i2s-controller + - mic-bias-gpios + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 GPIO_ACTIVE_HIGH>; + model = "wm5110"; + samsung,audio-routing = "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "RCV", "HPOUT3L", + "RCV", "HPOUT3R"; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000..84c4d6cba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,149 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with + secondary FIFO, s/w reset control and internal mux for root clock + source. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for + playback, stereo channel capture, secondary FIFO using internal + or external DMA, s/w reset control, internal mux for root clock + source and 7.1 channel TDM support for playback; TDM (Time division + multiplexing) is to allow transfer of multiple channel audio data on + single data line. + + samsung,exynos7-i2s: with all the available features of Exynos5 I2S. + Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO + with only external DMA and more number of root clock sampling + frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + assigned-clock-parents: true + assigned-clocks: true + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for I2S0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for I2S1 and I2S2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clock. I2S0 has internal mux to select the source + of root clock and I2S1 and I2S2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for I2S0 + - const: i2s_cdclk0 + - items: # for I2S1 + - const: i2s_cdclk1 + - items: # for I2S2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + interrupts: + maxItems: 1 + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/serial-midi.yaml b/Documentation/devicetree/bindings/sound/serial-midi.yaml new file mode 100644 index 000000000..4afc29376 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/serial-midi.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause + +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/serial-midi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Generic Serial MIDI Interface + +maintainers: + - Daniel Kaehn <kaehndan@gmail.com> + +description: + Generic MIDI interface using a serial device. This denotes that a serial device is + dedicated to MIDI communication, either to an external MIDI device through a DIN5 + or other connector, or to a known hardwired MIDI controller. This device must be a + child node of a serial node. + + Can only be set to use standard baud rates corresponding to supported rates of the + parent serial device. If the standard MIDI baud of 31.25 kBaud is needed + (as would be the case if interfacing with arbitrary external MIDI devices), + configure the clocks of the parent serial device so that a requested baud of 38.4 kBaud + resuts in the standard MIDI baud rate, and set the 'current-speed' property to 38400 (default) + +properties: + compatible: + const: serial-midi + + current-speed: + description: Baudrate to set the serial port to when this MIDI device is opened. + default: 38400 + +required: + - compatible + +additionalProperties: false + +examples: + - | + serial { + midi { + compatible = "serial-midi"; + }; + }; + - | + serial { + midi { + compatible = "serial-midi"; + current-speed = <115200>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml new file mode 100644 index 000000000..2bc7f00ce --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -0,0 +1,110 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale SGTL5000 Stereo Codec + +maintainers: + - Fabio Estevam <festevam@gmail.com> + +properties: + compatible: + const: fsl,sgtl5000 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + + clocks: + items: + - description: the clock provider of SYS_MCLK + + VDDA-supply: + description: the regulator provider of VDDA + + VDDIO-supply: + description: the regulator provider of VDDIO + + VDDD-supply: + description: the regulator provider of VDDD + + micbias-resistor-k-ohms: + description: The bias resistor to be used in kOhms. The resistor can take + values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not + mentioned or if the value is unknown, then micbias resistor is set to + 4k. + enum: [ 0, 2, 4, 8 ] + + micbias-voltage-m-volts: + description: The bias voltage to be used in mVolts. The voltage can take + values from 1.25V to 3V by 250mV steps. If this node is not mentioned + or the value is unknown, then the value is set to 1.25V. + $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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <®_3p3v>; + VDDIO-supply = <®_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml new file mode 100644 index 000000000..5428ba9e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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: + - $ref: name-prefix.yaml# + +properties: + compatible: + enum: + - dioo,dio2125 + - simple-audio-amplifier + + enable-gpios: + maxItems: 1 + + VCC-supply: + description: > + power supply for the device + + sound-name-prefix: true + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/meson8-gpio.h> + + analog-amplifier { + compatible = "simple-audio-amplifier"; + VCC-supply = <®ulator>; + enable-gpios = <&gpio GPIOH_3 0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000..b5fc35ee9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-mux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Multiplexer + +maintainers: + - Alexandre Belloni <aleandre.belloni@bootlin.com> + +description: | + Simple audio multiplexers are driven using gpios, allowing to select which of + their input line is connected to the output line. + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: simple-audio-mux + + mux-gpios: + description: | + GPIOs used to select the input line. + + sound-name-prefix: true + +required: + - compatible + - mux-gpios + +additionalProperties: false + +examples: + - | + mux { + compatible = "simple-audio-mux"; + mux-gpios = <&gpio 3 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000..ed19899bc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,500 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + + system-clock-fixed: + description: | + Specifies that the clock frequency should not be modified. + Implied when system-clock-frequency is specified, but can be used when + a clock is mapped to the device whose frequency cannot or should not be + changed. When mclk-fs is also specified, this restricts the device to a + single fixed sampling rate. + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + label: + maxItems: 1 + + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + $ref: "#/definitions/dai-tdm-slot-num" + dai-tdm-slot-width: + $ref: "#/definitions/dai-tdm-slot-width" + clocks: + maxItems: 1 + system-clock-frequency: + $ref: "#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "#/definitions/system-clock-direction-out" + system-clock-fixed: + $ref: "#/definitions/system-clock-fixed" + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": + const: 1 + "#size-cells": + const: 0 + + label: + $ref: "#/definitions/label" + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + $ref: "#/definitions/widgets" + simple-audio-card,routing: + $ref: "#/definitions/routing" + + # common properties + simple-audio-card,frame-master: + $ref: "#/definitions/frame-master" + simple-audio-card,bitclock-master: + $ref: "#/definitions/bitclock-master" + simple-audio-card,frame-inversion: + $ref: "#/definitions/frame-inversion" + simple-audio-card,bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + simple-audio-card,format: + $ref: "#/definitions/format" + simple-audio-card,mclk-fs: + $ref: "#/definitions/mclk-fs" + simple-audio-card,aux-devs: + $ref: "#/definitions/aux-devs" + simple-audio-card,convert-rate: + $ref: "#/definitions/convert-rate" + simple-audio-card,convert-channels: + $ref: "#/definitions/convert-channels" + simple-audio-card,prefix: + $ref: "#/definitions/prefix" + simple-audio-card,pin-switches: + $ref: "#/definitions/pin-switches" + simple-audio-card,hp-det-gpio: + maxItems: 1 + simple-audio-card,mic-det-gpio: + maxItems: 1 + +patternProperties: + "^simple-audio-card,cpu(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + maxItems: 1 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + + patternProperties: + "^cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + reg = <0>; + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + reg = <1>; + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + reg = <0>; + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + reg = <1>; + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu0 { + sound-dai = <&rcar_sound 1>; + }; + cpu1 { + sound-dai = <&rcar_sound 2>; + }; + cpu2 { + sound-dai = <&rcar_sound 3>; + }; + cpu3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + reg = <2>; + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-port.txt b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt new file mode 100644 index 000000000..1f66de3c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt @@ -0,0 +1,20 @@ +* SiRF SoC audio port + +Required properties: +- compatible: "sirf,audio-port" +- 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"). + +Example: + +audioport: audioport@b0040000 { + compatible = "sirf,audio-port"; + reg = <0xb0040000 0x10000>; + dmas = <&dmac1 3>, <&dmac1 8>; + dma-names = "rx", "tx"; +}; diff --git a/Documentation/devicetree/bindings/sound/sirf-audio.txt b/Documentation/devicetree/bindings/sound/sirf-audio.txt new file mode 100644 index 000000000..c88882ca3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-audio.txt @@ -0,0 +1,41 @@ +* SiRF atlas6 and prima2 internal audio codec and port based audio setups + +Required properties: +- compatible: "sirf,sirf-audio-card" +- sirf,audio-platform: phandle for the platform node +- sirf,audio-codec: phandle for the SiRF internal codec node + +Optional properties: +- hp-pa-gpios: Need to be present if the board need control external + headphone amplifier. +- spk-pa-gpios: Need to be present if the board need control external + speaker amplifier. +- hp-switch-gpios: Need to be present if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Ext Spk + * Line In + * Mic + +SiRF internal audio codec pins: + * HPOUTL + * HPOUTR + * SPKOUT + * Ext Mic + * Mic Bias + +Example: + +sound { + compatible = "sirf,sirf-audio-card"; + sirf,audio-codec = <&audiocodec>; + sirf,audio-platform = <&audioport>; + hp-pa-gpios = <&gpio 44 0>; + spk-pa-gpios = <&gpio 46 0>; + hp-switch-gpios = <&gpio 45 0>; +}; + 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 000000000..4b0795819 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml @@ -0,0 +1,94 @@ +# 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 + + 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: + description: Sampling rate reference clock + maxItems: 1 + + clock-names: + const: i2sclk + + resets: + maxItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + minItems: 1 + + dma-names: + items: + - const: tx + - const: rx + minItems: 1 + +if: + properties: + compatible: + contains: + const: canaan,k210-i2s + +then: + properties: + "#sound-dai-cells": + const: 1 + +else: + properties: + "#sound-dai-cells": + const: 0 + +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/soc-ac97link.txt b/Documentation/devicetree/bindings/sound/soc-ac97link.txt new file mode 100644 index 000000000..80152a87f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc-ac97link.txt @@ -0,0 +1,28 @@ +AC97 link bindings + +These bindings can be included within any other device node. + +Required properties: + - pinctrl-names: Has to contain following states to setup the correct + pinmuxing for the used gpios: + "ac97-running": AC97-link is active + "ac97-reset": AC97-link reset state + "ac97-warm-reset": AC97-link warm reset state + - ac97-gpios: List of gpio phandles with args in the order ac97-sync, + ac97-sdata, ac97-reset + + +Example: + +ssi { + ... + + pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset"; + pinctrl-0 = <&ac97link_running>; + pinctrl-1 = <&ac97link_running>; + pinctrl-2 = <&ac97link_reset>; + pinctrl-3 = <&ac97link_warm_reset>; + ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>; + + ... +}; 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 000000000..70f62ecd6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel@alsa-project.org> + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: "/schemas/types.yaml#/definitions/phandle" + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels that + depends on SoC. + ld11-aio,ld20-aio: + 0: hdmi + 1: pcmin2 + 2: line + 3: hpcmout1 + 4: pcmout3 + 5: hiecout1 + 6: epcmout2 + 7: epcmout3 + 8: hieccompout1 + pxs2-aio: + 0: hdmi + 1: line + 2: aux + 3: hiecout1 + 4: iecout1 + 5: hieccompout1 + 6: ieccompout1 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000..be6acfda9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel@alsa-project.org> + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels. + 0: line1 + 1: hp + 2: line2 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +additionalProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/sound-dai.yaml b/Documentation/devicetree/bindings/sound/sound-dai.yaml new file mode 100644 index 000000000..ff9036e43 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-dai.yaml @@ -0,0 +1,20 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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-receiver.txt b/Documentation/devicetree/bindings/sound/spdif-receiver.txt new file mode 100644 index 000000000..80f807bf8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/spdif-receiver.txt @@ -0,0 +1,10 @@ +Device-Tree bindings for dummy spdif receiver + +Required properties: + - compatible: should be "linux,spdif-dir". + +Example node: + + codec: spdif-receiver { + compatible = "linux,spdif-dir"; + }; diff --git a/Documentation/devicetree/bindings/sound/sprd-mcdt.txt b/Documentation/devicetree/bindings/sound/sprd-mcdt.txt new file mode 100644 index 000000000..274ba0acb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd-mcdt.txt @@ -0,0 +1,19 @@ +Spreadtrum Multi-Channel Data Transfer Binding + +The Multi-channel data transfer controller is used for sound stream +transmission between audio subsystem and other AP/CP subsystem. It +supports 10 DAC channel and 10 ADC channel, and each channel can be +configured with DMA mode or interrupt mode. + +Required properties: +- compatible: Should be "sprd,sc9860-mcdt". +- reg: Should contain registers address and length. +- interrupts: Should contain one interrupt shared by all channel. + +Example: + +mcdt@41490000 { + compatible = "sprd,sc9860-mcdt"; + reg = <0 0x41490000 0 0x170>; + interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt new file mode 100644 index 000000000..4b23e84b2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd-pcm.txt @@ -0,0 +1,23 @@ +* Spreadtrum DMA platfrom bindings + +Required properties: +- compatible: Should be "sprd,pcm-platform". +- dmas: Specify the list of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + +Example: + + audio_platform:platform@0 { + compatible = "sprd,pcm-platform"; + dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, + <&agcp_dma 3 3>, <&agcp_dma 4 4>, + <&agcp_dma 5 5>, <&agcp_dma 6 6>, + <&agcp_dma 7 7>, <&agcp_dma 8 8>, + <&agcp_dma 9 9>, <&agcp_dma 10 10>; + dma-names = "normal_p_l", "normal_p_r", + "normal_c_l", "normal_c_r", + "voice_c", "fast_p", + "loop_c", "loop_p", + "voip_c", "voip_p"; + }; diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt new file mode 100644 index 000000000..59381a778 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm2518.txt @@ -0,0 +1,20 @@ +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/ssm4567.txt b/Documentation/devicetree/bindings/sound/ssm4567.txt new file mode 100644 index 000000000..ec3d9e700 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm4567.txt @@ -0,0 +1,15 @@ +Analog Devices SSM4567 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm4567" + - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND), + 0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open). + +Example: + + ssm4567: ssm4567@34 { + compatible = "adi,ssm4567"; + reg = <0x34>; + }; diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt new file mode 100644 index 000000000..52265fb75 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -0,0 +1,101 @@ +STA32X audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta32x" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - Vdda-supply: regulator spec, providing 3.3V + - Vdd3-supply: regulator spec, providing 3.3V + - Vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is chosen. + This properties have to be specified as '/bits/ 8' values. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,fault-detect-recovery: + If present, fault detect recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,invalid-input-detect-mute: + If present, automatic invalid input detect mute is enabled. + +Example: + +codec: sta32x@38 { + compatible = "st,sta32x"; + reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 000000000..307398ef2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,131 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is chosen. + This properties have to be specified as '/bits/ 8' values. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If present, automatic invalid input detect mute is enabled. + + - st,activate-mute-output: + If present, a mute output will be activated in ase the volume will + reach a value lower than -76 dBFS. + + - st,bridge-immediate-off: + If present, the bridge will be switched off immediately after the + power-down-gpio goes low. Otherwise, the bridge will wait for 13 + million clock cycles to pass before shutting down. + + - st,noise-shape-dc-cut: + If present, the noise-shaping technique on the DC cutoff filter are + enabled. + + - st,powerdown-master-volume: + If present, the power-down pin and I2C power-down functions will + act on the master volume. Otherwise, the functions will act on the + mute commands. + + - st,powerdown-delay-divider: + If present, the bridge power-down time will be divided by the provided + value. If not specified, a divider of 1 will be used. Allowed values + are 1, 2, 4, 8, 16, 32, 64 and 128. + This property has to be specified as '/bits/ 8' value. + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt new file mode 100644 index 000000000..a6ffcdec6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -0,0 +1,164 @@ +STMicroelectronics sti ASoC cards + +The sti ASoC Sound Card can be used, for all sti SoCs using internal sti-sas +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.yaml. + +1) sti-uniperiph-dai: audio dai device. +--------------------------------------- + +Required properties: + - compatible: "st,stih407-uni-player-hdmi", "st,stih407-uni-player-pcm-out", + "st,stih407-uni-player-dac", "st,stih407-uni-player-spdif", + "st,stih407-uni-reader-pcm_in", "st,stih407-uni-reader-hdmi", + + - st,syscfg: phandle to boot-device system configuration registers + + - clock-names: name of the clocks listed in clocks property in the same order + + - reg: CPU DAI IP Base address and size entries, listed in same + order than the CPU_DAI properties. + + - reg-names: names of the mapped memory regions listed in regs property in + the same order. + + - interrupts: CPU_DAI interrupt line, listed in the same order than the + CPU_DAI properties. + + - dma: CPU_DAI DMA controller phandle and DMA request line, listed in the same + order than the CPU_DAI properties. + + - dma-names: identifier string for each DMA request line in the dmas property. + "tx" for "st,sti-uni-player" compatibility + "rx" for "st,sti-uni-reader" compatibility + +Required properties ("st,sti-uni-player" compatibility only): + - clocks: CPU_DAI IP clock source, listed in the same order than the + CPU_DAI properties. + +Optional properties: + - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for + external codecs connection. + + - pinctrl-names: should contain only one value - "default". + + - st,tdm-mode: to declare to set TDM mode for unireader and uniplayer IPs. + Only compartible with IPs in charge of the external I2S/TDM bus. + Should be declared depending on associated codec. + +Example: + + sti_uni_player1: sti-uni-player@8d81000 { + compatible = "st,stih407-uni-player-hdmi"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + dma-names = "tx"; + st,tdm-mode = <1>; + }; + + sti_uni_player2: sti-uni-player@8d82000 { + compatible = "st,stih407-uni-player-pcm-out"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_2>; + reg = <0x8D82000 0x158>; + interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>; + dmas = <&fdma0 4 0 1>; + dma-names = "tx"; + }; + + sti_uni_player3: sti-uni-player@8d85000 { + compatible = "st,stih407-uni-player-spdif"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_SPDIFF>; + reg = <0x8D85000 0x158>; + interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>; + dmas = <&fdma0 7 0 1>; + dma-names = "tx"; + }; + + sti_uni_reader1: sti-uni-reader@8d84000 { + compatible = "st,stih407-uni-reader-hdmi"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + reg = <0x8D84000 0x158>; + interrupts = <GIC_SPI 88 IRQ_TYPE_NONE>; + dmas = <&fdma0 6 0 1>; + dma-names = "rx"; + }; + +2) sti-sas-codec: internal audio codec IPs driver +------------------------------------------------- + +Required properties: + - compatible: "st,sti<chip>-sas-codec" . + Should be chip "st,stih416-sas-codec" or "st,stih407-sas-codec" + + - st,syscfg: phandle to boot-device system configuration registers. + + - pinctrl-0: SPDIF PIO description. + + - pinctrl-names: should contain only one value - "default". + +Example: + sti_sas_codec: sti-sas-codec { + compatible = "st,stih407-sas-codec"; + #sound-dai-cells = <1>; + st,reg_audio = <&syscfg_core>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdif_out >; + }; + +Example of audio card declaration: + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "sti audio card"; + + simple-audio-card,dai-link@0 { + /* DAC */ + format = "i2s"; + dai-tdm-slot-width = <32>; + cpu { + sound-dai = <&sti_uni_player2>; + }; + + codec { + sound-dai = <&sti_sasg_codec 1>; + }; + }; + simple-audio-card,dai-link@1 { + /* SPDIF */ + format = "left_j"; + cpu { + sound-dai = <&sti_uni_player3>; + }; + + codec { + sound-dai = <&sti_sasg_codec 0>; + }; + }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000..d3966ae04 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: i2sclk + - const: x8k + - const: x11k + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + "#clock-cells": + description: Configure the I2S device as MCLK clock provider. + const: 0 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000..56d206f97 --- /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 unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: phandle of the SAI sub-block + - description: index of the SAI sub-block + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + $ref: /schemas/types.yaml#/definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + 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 new file mode 100644 index 000000000..837e830c4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio data capture DMA + - description: IEC status bits capture DMA + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/storm.txt b/Documentation/devicetree/bindings/sound/storm.txt new file mode 100644 index 000000000..062a4c185 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/storm.txt @@ -0,0 +1,23 @@ +* Sound complex for Storm boards + +Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC +connected to a MAX98357A DAC via I2S. + +Required properties: + +- compatible : "google,storm-audio" +- cpu : Phandle of the CPU DAI +- codec : Phandle of the codec DAI + +Optional properties: + +- qcom,model : The user-visible name of this sound card. + +Example: + +sound { + compatible = "google,storm-audio"; + qcom,model = "ipq806x-storm"; + cpu = <&lpass_cpu>; + codec = <&max98357a>; +}; diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 000000000..a7eecad83 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,36 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + - compatible - One of: + "ti,tas2552" - TAS2552 + - reg - I2C slave address: it can be 0x40 if ADDR pin is 0 + or 0x41 if ADDR pin is 1. + - supply-*: Required supply regulators are: + "vbat" battery voltage + "iovdd" I/O Voltage + "avdd" Analog DAC Voltage + +Optional properties: + - enable-gpio - gpio pin to enable/disable the device + +tas2552 can receive its reference clock via MCLK, BCLK, IVCLKIN pin or use the +internal 1.8MHz. This CLKIN is used by the PLL. In addition to PLL, the PDM +reference clock is also selectable: PLL, IVCLKIN, BCLK or MCLK. +For system integration the dt-bindings/sound/tas2552.h header file provides +defined values to select and configure the PLL and PDM reference clocks. + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + vbat-supply = <®_vbat>; + iovdd-supply = <®_iovdd>; + avdd-supply = <®_avdd>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +https://www.ti.com/product/TAS2552 diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000..cb519a4b6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,80 @@ +# 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/tas2563 + https://www.ti.com/lit/gpn/tas2564 + https://www.ti.com/lit/gpn/tas2110 + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + - ti,tas2564 + - ti,tas2110 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000..1859fbe1c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704@thundersoft.com> + +description: | + The TAS2770 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x41 to 0x48. + + reset-gpio: + maxItems: 1 + description: GPIO used to reset the device. + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@41 { + compatible = "ti,tas2770"; + reg = <0x41>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 14 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml new file mode 100644 index 000000000..079cb6f8d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020-2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas27xx.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2764/TAS2780 Smart PA + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated speaker voltage and current sense provides + for real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2764 + - ti,tas2780 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + '#sound-dai-cells': + # 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 + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <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 new file mode 100644 index 000000000..7c8fd37c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -0,0 +1,48 @@ +Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers + +The codec is controlled through an I2C interface. It also has two other +signals that can be wired up to GPIOs: reset (strongly recommended), and +powerdown (optional). + +Required properties: + +- compatible: should be one of the following: + - "ti,tas5707" + - "ti,tas5711", + - "ti,tas5717", + - "ti,tas5719", + - "ti,tas5721" +- reg: The I2C address of the device +- #sound-dai-cells: must be equal to 0 + +Optional properties: + +- reset-gpios: GPIO specifier for the TAS571x's active low reset line +- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line +- clocks: clock phandle for the MCLK input +- clock-names: should be "mclk" +- AVDD-supply: regulator phandle for the AVDD supply (all chips) +- DVDD-supply: regulator phandle for the DVDD supply (all chips) +- HPVDD-supply: regulator phandle for the HPVDD supply (5717/5719) +- PVDD_AB-supply: regulator phandle for the PVDD_AB supply (5717/5719) +- PVDD_CD-supply: regulator phandle for the PVDD_CD supply (5717/5719) +- PVDD_A-supply: regulator phandle for the PVDD_A supply (5711) +- PVDD_B-supply: regulator phandle for the PVDD_B supply (5711) +- PVDD_C-supply: regulator phandle for the PVDD_C supply (5711) +- PVDD_D-supply: regulator phandle for the PVDD_D supply (5711) +- DRVDD-supply: regulator phandle for the DRVDD supply (5721) +- PVDD-supply: regulator phandle for the PVDD supply (5721) + +Example: + + tas5717: audio-codec@2a { + compatible = "ti,tas5717"; + reg = <0x2a>; + #sound-dai-cells = <0>; + + reset-gpios = <&gpio5 1 GPIO_ACTIVE_LOW>; + pdn-gpios = <&gpio5 2 GPIO_ACTIVE_LOW>; + + clocks = <&clk_core CLK_I2S>; + clock-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt new file mode 100644 index 000000000..df99ca945 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -0,0 +1,26 @@ +Texas Instruments TAS5720 Mono Audio amplifier + +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: + +https://www.ti.com/product/TAS5720L +https://www.ti.com/product/TAS5720M +https://www.ti.com/product/TAS5722L + +Required properties: + +- compatible : "ti,tas5720", + "ti,tas5722" +- reg : I2C slave address +- dvdd-supply : phandle to a 3.3-V supply for the digital circuitry +- pvdd-supply : phandle to a supply used for the Class-D amp and the analog + +Example: + +tas5720: tas5720@6c { + compatible = "ti,tas5720"; + reg = <0x6c>; + dvdd-supply = <&vdd_3v3_reg>; + pvdd-supply = <&_supply_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/tas5805m.yaml b/Documentation/devicetree/bindings/sound/tas5805m.yaml new file mode 100644 index 000000000..3aade02d8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas5805m.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tas5805m.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: TAS5805M audio amplifier + +maintainers: + - Daniel Beer <daniel.beer@igorinstitute.com> + +description: | + The TAS5805M is a class D audio amplifier with a built-in DSP. + +properties: + compatible: + enum: + - ti,tas5805m + + reg: + maxItems: 1 + description: | + I2C address of the amplifier. See the datasheet for possible values. + + pvdd-supply: + description: | + Regulator for audio power supply (PVDD in the datasheet). + + pdn-gpios: + description: | + Power-down control GPIO (PDN pin in the datasheet). + + ti,dsp-config-name: + description: | + The name of the DSP configuration that should be loaded for this + instance. Configuration blobs are sequences of register writes + generated from TI's PPC3 tool. + $ref: /schemas/types.yaml#/definitions/string + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + tas5805m: tas5805m@2c { + reg = <0x2c>; + compatible = "ti,tas5805m"; + + pvdd-supply = <&audiopwr>; + pdn-gpios = <&tlmm 160 0>; + + ti,dsp-config-name = "mono_pbtl_48khz"; + }; + }; + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/tda7419.txt b/Documentation/devicetree/bindings/sound/tda7419.txt new file mode 100644 index 000000000..6b85ec38d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tda7419.txt @@ -0,0 +1,38 @@ +TDA7419 audio processor + +This device supports I2C only. + +Required properties: + +- compatible : "st,tda7419" +- reg : the I2C address of the device. +- vdd-supply : a regulator spec for the common power supply (8-10V) + +Optional properties: + +- st,mute-gpios : a GPIO spec for the MUTE pin. + +Pins on the device (for linking into audio routes): + + * SE3L + * SE3R + * SE2L + * SE2R + * SE1L + * SE1R + * DIFFL + * DIFFR + * MIX + * OUTLF + * OUTRF + * OUTLR + * OUTRR + * OUTSW + +Example: + +ap: tda7419@44 { + compatible = "st,tda7419"; + reg = <0x44>; + vdd-supply = <&vdd_9v0_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt new file mode 100644 index 000000000..4bb513ae6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -0,0 +1,29 @@ +TDM slot: + +This specifies audio DAI's TDM slot. + +TDM slot properties: +dai-tdm-slot-num : Number of slots in use. +dai-tdm-slot-width : Width in bits for each slot. +dai-tdm-slot-tx-mask : Transmit direction slot mask, optional +dai-tdm-slot-rx-mask : Receive direction slot mask, optional + +For instance: + dai-tdm-slot-num = <2>; + dai-tdm-slot-width = <8>; + dai-tdm-slot-tx-mask = <0 1>; + dai-tdm-slot-rx-mask = <1 0>; + +And for each specified driver, there could be one .of_xlate_tdm_slot_mask() +to specify an explicit mapping of the channels and the slots. If it's absent +the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the +tx and rx masks. + +For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit +for an active slot as default, and the default active bits are at the LSB of +the masks. + +The explicit masks are given as array of integers, where the first +number presents bit-0 (LSB), second presents bit-1, etc. Any non zero +number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask() +does not do anything, if either mask is set non zero value. diff --git a/Documentation/devicetree/bindings/sound/test-component.yaml b/Documentation/devicetree/bindings/sound/test-component.yaml new file mode 100644 index 000000000..9c40a2122 --- /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 new file mode 100644 index 000000000..1620e6848 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tfa9879.txt @@ -0,0 +1,23 @@ +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,ads117x.txt b/Documentation/devicetree/bindings/sound/ti,ads117x.txt new file mode 100644 index 000000000..7db19b508 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,ads117x.txt @@ -0,0 +1,11 @@ +Texas Intstruments ADS117x ADC + +Required properties: + + - compatible : "ti,ads1174" or "ti,ads1178" + +Example: + +ads1178 { + compatible = "ti,ads1178"; +}; 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 000000000..20ea5883b --- /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 000000000..859d369c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi <peter.ujfalusi@ti.com> +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments J721e Common Processor Board Audio Support + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: | + The Infotainment board plugs into the Common Processor Board, the support of the + extension board is extending the CPB audio support, decribed in: + sound/ti,j721e-cpb-audio.txt + + The audio support on the Infotainment Expansion Board consists of McASP0 + connected to two pcm3168a codecs with dedicated set of serializers to each. + The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin. + + In order to support 48KHz and 44.1KHz family of sampling rates the parent clock + for AUDIO_REFCLK0 needs to be changed between PLL4 (for 48KHz) and PLL15 (for + 44.1KHz). The same PLLs are used for McASP0's AUXCLK clock via different + HSDIVIDER. + + Note: the same PLL4 and PLL15 is used by the audio support on the CPB! + + Clocking setup for 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + | |-> MCASP0_AUXCLK ---> McASP0.auxclk + | + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI + + Clocking setup for 44.1KHz family: + PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + | |-> MCASP0_AUXCLK ---> McASP0.auxclk + | + |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI + +properties: + compatible: + items: + - const: ti,j721e-cpb-ivi-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + ti,cpb-mcasp: + description: phandle to McASP used on CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,cpb-codec: + description: phandle to the pcm3168a codec used on the CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-mcasp: + description: phandle to McASP used on IVI + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-codec-a: + description: phandle to the pcm3168a-A codec on the expansion board + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-codec-b: + description: phandle to the pcm3168a-B codec on the expansion board + $ref: /schemas/types.yaml#/definitions/phandle + + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: Parent for CPB_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + - description: Parent for CPB_SCKI clock (for 44.1KHz) + - description: AUXCLK clock for McASP used by IVI audio + - description: Parent for IVI_McASP auxclk (for 48KHz) + - description: Parent for IVI_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on IVI + - description: Parent for IVI_SCKI clock (for 48KHz) + - description: Parent for IVI_SCKI clock (for 44.1KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-mcasp-auxclk-44100 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + - const: cpb-codec-scki-44100 + - const: ivi-mcasp-auxclk + - const: ivi-mcasp-auxclk-48000 + - const: ivi-mcasp-auxclk-44100 + - const: ivi-codec-scki + - const: ivi-codec-scki-48000 + - const: ivi-codec-scki-44100 + +required: + - compatible + - model + - ti,cpb-mcasp + - ti,cpb-codec + - ti,ivi-mcasp + - ti,ivi-codec-a + - ti,ivi-codec-b + - clocks + - clock-names + +additionalProperties: false + +examples: + - |+ + sound { + compatible = "ti,j721e-cpb-ivi-audio"; + model = "j721e-cpb-ivi"; + + ti,cpb-mcasp = <&mcasp10>; + ti,cpb-codec = <&pcm3168a_1>; + + ti,ivi-mcasp = <&mcasp0>; + ti,ivi-codec-a = <&pcm3168a_a>; + ti,ivi-codec-b = <&pcm3168a_b>; + + clocks = <&k3_clks 184 1>, + <&k3_clks 184 2>, <&k3_clks 184 4>, + <&k3_clks 157 371>, + <&k3_clks 157 400>, <&k3_clks 157 401>, + <&k3_clks 174 1>, + <&k3_clks 174 2>, <&k3_clks 174 4>, + <&k3_clks 157 301>, + <&k3_clks 157 330>, <&k3_clks 157 331>; + clock-names = "cpb-mcasp-auxclk", + "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100", + "cpb-codec-scki", + "cpb-codec-scki-48000", "cpb-codec-scki-44100", + "ivi-mcasp-auxclk", + "ivi-mcasp-auxclk-48000", "ivi-mcasp-auxclk-44100", + "ivi-codec-scki", + "ivi-codec-scki-48000", "ivi-codec-scki-44100"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 000000000..4df17185a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt new file mode 100644 index 000000000..a02ecaab5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt @@ -0,0 +1,56 @@ +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,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000..9681b72b4 --- /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 Device Tree Bindings + +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: name-prefix.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,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 000000000..234dad296 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,48 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + + - ti,mid-z-channel-X: Boolean properties, X being a number from 1 to 6. + If given, channel X will start with the Mid-Z start + sequence, otherwise the default Low-Z scheme is used. + + The correct configuration depends on how the power + stages connected to the PWM output pins work. Not all + power stages are compatible to Mid-Z - please refer + to the datasheets for more details. + + Most systems should not set any of these properties. + + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt new file mode 100644 index 000000000..00940c489 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt @@ -0,0 +1,22 @@ +Texas Instruments TAS6424 Quad-Channel Audio amplifier + +The TAS6424 serial control bus communicates through I2C protocols. + +Required properties: + - compatible: "ti,tas6424" - TAS6424 + - reg: I2C slave address + - sound-dai-cells: must be equal to 0 + - standby-gpios: GPIO used to shut the TAS6424 down. + - mute-gpios: GPIO used to mute all the outputs + +Example: + +tas6424: tas6424@6a { + compatible = "ti,tas6424"; + reg = <0x6a>; + + #sound-dai-cells = <0>; +}; + +For more product information please see the link below: +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 000000000..83936f594 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw@axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: audio-codec@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml b/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml new file mode 100644 index 000000000..785930658 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Intended MICBIAS voltage (datasheet section 9.6.7). + enum: + - 0 # 2.1 V + - 1 # 2.2 V + - 2 # 2.3 V + - 3 # 2.4 V + - 4 # 2.5 V + - 5 # 2.6 V + - 6 # 2.7 V + - 7 # 2.8 V + default: 1 + + ti,debounce-release-ms: + description: key release debounce time in ms (datasheet section 9.6.7). + enum: + - 0 + - 20 + default: 20 + + ti,debounce-press-ms: + description: key press debounce time in ms (datasheet section 9.6.7). + enum: + - 2 + - 40 + - 80 + - 120 + default: 80 + + ti,debounce-insertion-ms: + description: headset insertion debounce time in ms (datasheet section 9.6.5). + enum: + - 2 + - 30 + - 60 + - 90 + - 120 + - 150 + - 1000 + - 2000 + default: 90 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: audio-controller@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio1>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml new file mode 100644 index 000000000..ee6986148 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -0,0 +1,209 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter + +maintainers: + - Andrew Davis <afd@ti.com> + +description: | + The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital + PDM microphones recording), high-performance audio, analog-to-digital + converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140 + family supports line and microphone Inputs, and offers a programmable + microphone bias or supply voltage generation. + + Specifications can be found at: + https://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + +properties: + compatible: + enum: + - ti,tlv320adc3140 + - ti,tlv320adc5140 + - ti,tlv320adc6140 + + reg: + maxItems: 1 + description: | + I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpios: + maxItems: 1 + description: | + GPIO used for hardware reset. + + areg-supply: + description: | + Regulator with AVDD at 3.3V. If not defined then the internal regulator + is enabled. + + ti,mic-bias-source: + description: | + Indicates the source for MIC Bias. + 0 - Mic bias is set to VREF + 1 - Mic bias is set to VREF × 1.096 + 6 - Mic bias is set to AVDD + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 6] + + ti,vref-source: + description: | + Indicates the source for MIC Bias. + 0 - Set VREF to 2.75V + 1 - Set VREF to 2.5V + 2 - Set VREF to 1.375V + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + + ti,pdm-edge-select: + description: | + Defines the PDMCLK sampling edge configuration for the PDM inputs. This + array is defined as <PDMIN1 PDMIN2 PDMIN3 PDMIN4>. + + 0 - (default) Odd channel is latched on the negative edge and even + channel is latched on the positive edge. + 1 - Odd channel is latched on the positive edge and even channel is + 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 + PDMIN3 - PDMCLK latching edge used for channel 5 and 6 data + PDMIN4 - PDMCLK latching edge used for channel 7 and 8 data + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + maximum: 1 + default: [0, 0, 0, 0] + + ti,gpi-config: + description: | + Defines the configuration for the general purpose input pins (GPI). + The array is defined as <GPI1 GPI2 GPI3 GPI4>. + + 0 - (default) disabled + 1 - GPIX is configured as a general-purpose input (GPI) + 2 - GPIX is configured as a master clock input (MCLK) + 3 - GPIX is configured as an ASI input for daisy-chain (SDIN) + 4 - GPIX is configured as a PDM data input for channel 1 and channel + (PDMDIN1) + 5 - GPIX is configured as a PDM data input for channel 3 and channel + (PDMDIN2) + 6 - GPIX is configured as a PDM data input for channel 5 and channel + (PDMDIN3) + 7 - GPIX is configured as a PDM data input for channel 7 and channel + (PDMDIN4) + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + maximum: 7 + default: [0, 0, 0, 0] + + ti,asi-tx-drive: + type: boolean + description: | + When set the device will set the Tx ASI output to a Hi-Z state for unused + data cycles. Default is to drive the output low on unused ASI cycles. + +patternProperties: + '^ti,gpo-config-[1-4]$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + Defines the configuration and output driver for the general purpose + output pins (GPO). These values are pairs, the first value is for the + configuration type and the second value is for the output drive type. + The array is defined as <GPO_CFG GPO_DRV> + + GPO output configuration can be one of the following: + + 0 - (default) disabled + 1 - GPOX is configured as a general-purpose output (GPO) + 2 - GPOX is configured as a device interrupt output (IRQ) + 3 - GPOX is configured as a secondary ASI output (SDOUT2) + 4 - GPOX is configured as a PDM clock output (PDMCLK) + + GPO output drive configuration for the GPO pins can be one of the following: + + 0d - (default) Hi-Z output + 1d - Drive active low and active high + 2d - Drive active low and weak high + 3d - Drive active low and Hi-Z + 4d - Drive weak low and active high + 5d - Drive Hi-Z and active high + + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tlv320adc5140"; + reg = <0x4c>; + ti,mic-bias-source = <6>; + ti,pdm-edge-select = <0 1 0 1>; + ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; + ti,gpo-config-1 = <0 0>; + ti,gpo-config-2 = <0 0>; + reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 000000000..bbad98d5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,77 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + +- reg - <int> - I2C slave address +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + + +Optional properties: + +- reset-gpios - GPIO specification for the active low RESET input. +- ai31xx-micbias-vg - MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- ai31xx-ocmv - output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + +Deprecated properties: + +- gpio-reset - gpio pin number used for codec reset + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + +CODEC input pins: + * MIC1LP, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + +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.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt new file mode 100644 index 000000000..f59125bc7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -0,0 +1,42 @@ +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 + "ti,tas2505" TAS2505, TAS2521 + - 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 new file mode 100644 index 000000000..20931a63f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -0,0 +1,97 @@ +Texas Instruments - tlv320aic3x Codec module + +The tlv320aic3x serial control bus communicates through both I2C and SPI bus 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. + +I2C 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>; +}; + +SPI example: + +spi0: spi@f0000000 { + tlv320aic3x: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + spi-max-frequency = <1000000>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + ai3x-ocmv = <0>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt new file mode 100644 index 000000000..6dfa740e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tpa6130a2.txt @@ -0,0 +1,27 @@ +Texas Instruments - tpa6130a2 Codec module + +The tpa6130a2 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tpa6130a2" - TPA6130A2 + "ti,tpa6140a2" - TPA6140A2 + + +- reg - <int> - I2C slave address + +- Vdd-supply - <phandle> - power supply regulator + +Optional properties: + +- power-gpio - gpio pin to power the device + +Example: + +tpa6130a2: tpa6130a2@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt new file mode 100644 index 000000000..7eea32e9d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -0,0 +1,22 @@ +TSCS42XX Audio CODEC + +Required Properties: + + - compatible : "tempo,tscs42A1" for analog mic + "tempo,tscs42A2" for digital mic + + - reg : <0x71> for analog mic + <0x69> for digital mic + + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + +Example: + +wookie: codec@69 { + compatible = "tempo,tscs42A2"; + reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; +}; diff --git a/Documentation/devicetree/bindings/sound/tscs454.txt b/Documentation/devicetree/bindings/sound/tscs454.txt new file mode 100644 index 000000000..3ba3e2d2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tscs454.txt @@ -0,0 +1,23 @@ +TSCS454 Audio CODEC + +Required Properties: + + - compatible : "tempo,tscs454" + + - reg : <0x69> + + - clock-names: Must one of the following "xtal", "mclk1", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + + Note: If clock is not provided then bit clock is assumed + +Example: + +redwood: codec@69 { + #sound-dai-cells = <1>; + compatible = "tempo,tscs454"; + reg = <0x69>; + clock-names = "mclk1"; + clocks = <&audio_mclk>; +}; diff --git a/Documentation/devicetree/bindings/sound/uda1334.txt b/Documentation/devicetree/bindings/sound/uda1334.txt new file mode 100644 index 000000000..f64071b25 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/uda1334.txt @@ -0,0 +1,17 @@ +UDA1334 audio CODEC + +This device uses simple GPIO pins for controlling codec settings. + +Required properties: + + - compatible : "nxp,uda1334" + - nxp,mute-gpios: a GPIO spec for the MUTE pin. + - nxp,deemph-gpios: a GPIO spec for the De-emphasis pin + +Example: + +uda1334: audio-codec { + compatible = "nxp,uda1334"; + nxp,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + nxp,deemph-gpios = <&gpio3 3 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/devicetree/bindings/sound/ux500-mop500.txt b/Documentation/devicetree/bindings/sound/ux500-mop500.txt new file mode 100644 index 000000000..48e071c96 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-mop500.txt @@ -0,0 +1,39 @@ +* MOP500 Audio Machine Driver + +This node is responsible for linking together all ux500 Audio Driver components. + +Required properties: + - compatible : "stericsson,snd-soc-mop500" + +Non-standard properties: + - stericsson,cpu-dai : Phandle to the CPU-side DAI + - stericsson,audio-codec : Phandle to the Audio CODEC + - stericsson,card-name : Over-ride default card name + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; diff --git a/Documentation/devicetree/bindings/sound/ux500-msp.txt b/Documentation/devicetree/bindings/sound/ux500-msp.txt new file mode 100644 index 000000000..7dd1b9616 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-msp.txt @@ -0,0 +1,42 @@ +* ux500 MSP (CPU-side Digital Audio Interface) + +Required properties: + - compatible :"stericsson,ux500-msp-i2s" + - reg : Physical base address and length of the device's registers. + +Optional properties: + - interrupts : The interrupt output from the device. + - <name>-supply : Phandle to the regulator <name> supply + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,platform-pcm-dma = <&pcm>; + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + pcm: ux500-pcm { + compatible = "stericsson,ux500-pcm"; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; diff --git a/Documentation/devicetree/bindings/sound/widgets.txt b/Documentation/devicetree/bindings/sound/widgets.txt new file mode 100644 index 000000000..b6de5ba3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/widgets.txt @@ -0,0 +1,20 @@ +Widgets: + +This mainly specifies audio off-codec DAPM widgets. + +Each entry is a pair of strings in DT: + + "template-wname", "user-supplied-wname" + +The "template-wname" being the template widget name and currently includes: +"Microphone", "Line", "Headphone" and "Speaker". + +The "user-supplied-wname" being the user specified widget name. + +For instance: + simple-audio-widgets = + "Microphone", "Microphone Jack", + "Line", "Line In Jack", + "Line", "Line Out Jack", + "Headphone", "Headphone Jack", + "Speaker", "Speaker External"; diff --git a/Documentation/devicetree/bindings/sound/wlf,arizona.yaml b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml new file mode 100644 index 000000000..1627c0bb6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,arizona.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic/Wolfson Microelectronics Arizona class audio SoCs + +maintainers: + - patches@opensource.cirrus.com + +description: | + These devices are audio SoCs with extensive digital capabilities and a range + of analogue I/O. + + This document lists sound specific bindings, see the primary binding + document ../mfd/arizona.yaml + +properties: + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + wlf,inmode: + description: + A list of INn_MODE register values, where n is the number of input + signals. Valid values are 0 (Differential), 1 (Single-ended) and + 2 (Digital Microphone). If absent, INn_MODE registers set to 0 by + default. If present, values must be specified less than or equal + to the number of input signals. If values less than the number of + input signals, elements that have not been specified are set to 0 by + default. Entries are <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, + wm8997) and <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814) + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 2 + default: 0 + + wlf,out-mono: + description: + A list of boolean values indicating whether each output is mono + or stereo. Position within the list indicates the output affected + (eg. First entry in the list corresponds to output 1). A non-zero + value indicates a mono output. If present, the number of values + should be less than or equal to the number of outputs, if less values + are supplied the additional outputs will be treated as stereo. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 6 + items: + minimum: 0 + maximum: 1 + default: 0 + + wlf,dmic-ref: + description: + DMIC reference voltage source for each input, can be selected from + either MICVDD or one of the MICBIAS's, defines (ARIZONA_DMIC_xxxx) + are provided in dt-bindings/mfd/arizona.h. If present, the number + of values should be less than or equal to the number of inputs, + unspecified inputs will use the chip default. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 3 + default: 0 + + wlf,max-channels-clocked: + description: + The maximum number of channels to be clocked on each AIF, useful for + I2S systems with multiple data lines being mastered. Specify one + cell for each AIF to be configured, specify zero for AIFs that should + be handled normally. If present, number of cells must be less than + or equal to the number of AIFs. If less than the number of AIFs, for + cells that have not been specified the corresponding AIFs will be + treated as default setting. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 3 + items: + default: 0 + + wlf,spk-fmt: + description: + PDM speaker data format, must contain 2 cells (OUT5 and OUT6). See + the datasheet for values. The second cell is ignored for codecs that + do not have OUT6 (wm5102, wm8997, wm8998, wm1814) + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + wlf,spk-mute: + description: + PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6). See + the datasheet for values. The second cell is ignored for codecs that + do not have OUT6 (wm5102, wm8997, wm8998, wm1814) + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + wlf,out-volume-limit: + description: + The volume limit value that should be applied to each output + channel. See the datasheet for exact values. Channels are specified + in the order OUT1L, OUT1R, OUT2L, OUT2R, etc. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 12 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml new file mode 100644 index 000000000..15795f63b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + wm8731_i2c: codec@0 { + compatible = "wlf,wm8731"; + reg = <0>; + spi-max-frequency = <12500000>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; + - | + + i2c { + #address-cells = <1>; + #size-cells = <0>; + wm8731_spi: codec@1b { + compatible = "wlf,wm8731"; + reg = <0x1b>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000..7105ed5fd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/wlf,wm8903.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: WM8903 audio codec + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8903 + + reg: + maxItems: 1 + + gpio-controller: true + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + micdet-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + description: Default register value for R6 (Mic Bias). + + micdet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 100 + description: The debounce delay for microphone detection in mS. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml new file mode 100644 index 000000000..7386abb3a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8940.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8940 Codec + +maintainers: + - patches@opensource.cirrus.com + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8940 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml new file mode 100644 index 000000000..5e172e946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml @@ -0,0 +1,121 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8962.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8962 Ultra-Low Power Stereo CODEC + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8962 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + interrupts: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + AVDD-supply: + description: Analogue supply. + + CPVDD-supply: + description: Charge pump power supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + MICVDD-supply: + description: Microphone bias amp supply. + + PLLVDD-supply: + description: PLL Supply + + SPKVDD1-supply: + description: Supply for left speaker drivers. + + SPKVDD2-supply: + description: Supply for right speaker drivers. + + spk-mono: + $ref: /schemas/types.yaml#/definitions/flag + description: + If present, the SPK_MONO bit of R51 (Class D Control 2) gets set, + indicating that the speaker is in mono mode. + + mic-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Default register value for R48 (Additional Control 4). + If absent, the default should be the register default. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + description: + A list of GPIO configuration register values. If absent, no + configuration of these registers is performed. Note that only values + within [0x0, 0xffff] are valid. Any other value is regarded as setting + the GPIO register to its reset value 0x0. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + - PLLVDD-supply + - SPKVDD1-supply + - SPKVDD2-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/imx6qdl-clock.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8962: codec@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; + clocks = <&clks IMX6QDL_CLK_CKO>; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + CPVDD-supply = <®_audio>; + MICVDD-supply = <®_audio>; + PLLVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; + gpio-cfg = < + 0x0000 /* 0:Default */ + 0x0000 /* 1:Default */ + 0x0013 /* 2:FN_DMICCLK */ + 0x0000 /* 3:Default */ + 0x8014 /* 4:FN_DMICCDAT */ + 0x0000 /* 5:Default */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt new file mode 100644 index 000000000..01d3a7c83 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt @@ -0,0 +1,15 @@ +WM8974 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "wlf,wm8974" + - reg: the I2C address or SPI chip select number of the device + +Examples: + +codec: wm8974@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000..1c8985d4d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec + +maintainers: + - patches@opensource.cirrus.com + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wm8510.txt b/Documentation/devicetree/bindings/sound/wm8510.txt new file mode 100644 index 000000000..e6b6cc041 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8510.txt @@ -0,0 +1,18 @@ +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 new file mode 100644 index 000000000..f3a6485f4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8523.txt @@ -0,0 +1,16 @@ +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 new file mode 100644 index 000000000..f6c0c263b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8524.txt @@ -0,0 +1,16 @@ +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 new file mode 100644 index 000000000..ff3f9f5f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8580.txt @@ -0,0 +1,16 @@ +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 new file mode 100644 index 000000000..c30a1387c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8711.txt @@ -0,0 +1,18 @@ +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 new file mode 100644 index 000000000..a3608b4c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8728.txt @@ -0,0 +1,18 @@ +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/wm8737.txt b/Documentation/devicetree/bindings/sound/wm8737.txt new file mode 100644 index 000000000..eda1ec6a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8737.txt @@ -0,0 +1,18 @@ +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/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt new file mode 100644 index 000000000..b69e196c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -0,0 +1,29 @@ +WM8741 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8741" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + +Example: + +wm8741: codec@1a { + compatible = "wlf,wm8741"; + reg = <0x1a>; + + diff-mode = <3>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8750.yaml b/Documentation/devicetree/bindings/sound/wm8750.yaml new file mode 100644 index 000000000..24246ac7b --- /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 new file mode 100644 index 000000000..eca9e5a82 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8753.txt @@ -0,0 +1,40 @@ +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/wm8770.txt b/Documentation/devicetree/bindings/sound/wm8770.txt new file mode 100644 index 000000000..cac762a11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8770.txt @@ -0,0 +1,16 @@ +WM8770 audio CODEC + +This device supports SPI. + +Required properties: + + - compatible : "wlf,wm8770" + + - reg : the chip select number. + +Example: + +wm8770: codec@1 { + compatible = "wlf,wm8770"; + reg = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt new file mode 100644 index 000000000..01173369c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8776.txt @@ -0,0 +1,18 @@ +WM8776 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8776" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +wm8776: codec@1a { + compatible = "wlf,wm8776"; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8782.txt b/Documentation/devicetree/bindings/sound/wm8782.txt new file mode 100644 index 000000000..256cdec6e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8782.txt @@ -0,0 +1,17 @@ +WM8782 stereo ADC + +This device does not have any control interface or reset pins. + +Required properties: + + - compatible : "wlf,wm8782" + - 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) + +Example: + +wm8782: stereo-adc { + compatible = "wlf,wm8782"; + Vdda-supply = <&vdda_supply>; + Vdd-supply = <&vdd_supply>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8804.txt b/Documentation/devicetree/bindings/sound/wm8804.txt new file mode 100644 index 000000000..2c1641c17 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8804.txt @@ -0,0 +1,25 @@ +WM8804 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8804" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + - PVDD-supply, DVDD-supply : Power supplies for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + + - wlf,reset-gpio: A GPIO specifier for the GPIO controlling the reset pin + +Example: + +wm8804: codec@1a { + compatible = "wlf,wm8804"; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 000000000..66bf26142 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,33 @@ +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 new file mode 100644 index 000000000..85d3b2871 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -0,0 +1,42 @@ +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. + + - wlf,hp-cfg: A list of headphone jack detect configuration register values. + The list must be 3 entries long. + hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4). + hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2). + hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1). + + - wlf,gpio-cfg: A list of GPIO configuration register values. + The list must be 2 entries long. + gpio-cfg[0]: ALRCGPIO of R9 (Audio interface) + gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4). + +Example: + +wm8960: codec@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + wlf,shared-lrclk; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt new file mode 100644 index 000000000..8fa947509 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -0,0 +1,112 @@ +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. + +Pins on the device (for linking into audio routes): + + * IN1LN + * IN1LP + * IN2LN + * IN2LP:VXRN + * IN1RN + * IN1RP + * IN2RN + * IN2RP:VXRP + * SPKOUTLP + * SPKOUTLN + * SPKOUTRP + * SPKOUTRN + * HPOUT1L + * HPOUT1R + * HPOUT2P + * HPOUT2N + * LINEOUT1P + * LINEOUT1N + * LINEOUT2P + * LINEOUT2N + +Example: + +wm8994: codec@1a { + 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/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000..cbc93c8f4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/xlnx,i2s.txt b/Documentation/devicetree/bindings/sound/xlnx,i2s.txt new file mode 100644 index 000000000..5e7c7d5bb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,i2s.txt @@ -0,0 +1,28 @@ +Device-Tree bindings for Xilinx I2S PL block + +The IP supports I2S based playback/capture audio + +Required property: + - compatible: "xlnx,i2s-transmitter-1.0" for playback and + "xlnx,i2s-receiver-1.0" for capture + +Required property common to both I2S playback and capture: + - reg: Base address and size of the IP core instance. + - xlnx,dwidth: sample data width. Can be any of 16, 24. + - xlnx,num-channels: Number of I2S streams. Can be any of 1, 2, 3, 4. + supported channels = 2 * xlnx,num-channels + +Example: + + i2s_receiver@a0080000 { + compatible = "xlnx,i2s-receiver-1.0"; + reg = <0x0 0xa0080000 0x0 0x10000>; + xlnx,dwidth = <0x18>; + xlnx,num-channels = <1>; + }; + i2s_transmitter@a0090000 { + compatible = "xlnx,i2s-transmitter-1.0"; + reg = <0x0 0xa0090000 0x0 0x10000>; + xlnx,dwidth = <0x18>; + xlnx,num-channels = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/xlnx,spdif.txt b/Documentation/devicetree/bindings/sound/xlnx,spdif.txt new file mode 100644 index 000000000..15c2d64d2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,spdif.txt @@ -0,0 +1,28 @@ +Device-Tree bindings for Xilinx SPDIF IP + +The IP supports playback and capture of SPDIF audio + +Required properties: + - compatible: "xlnx,spdif-2.0" + - clock-names: List of input clocks. + Required elements: "s_axi_aclk", "aud_clk_i" + - clocks: Input clock specifier. Refer to common clock bindings. + - reg: Base address and address length of the IP core instance. + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - xlnx,spdif-mode: 0 :- receiver mode + 1 :- transmitter mode + - xlnx,aud_clk_i: input audio clock value. + +Example: + spdif_0: spdif@80010000 { + clock-names = "aud_clk_i", "s_axi_aclk"; + clocks = <&misc_clk_0>, <&clk 71>; + compatible = "xlnx,spdif-2.0"; + interrupt-names = "spdif_interrupt"; + interrupt-parent = <&gic>; + interrupts = <0 91 4>; + reg = <0x0 0x80010000 0x0 0x10000>; + xlnx,spdif-mode = <1>; + xlnx,aud_clk_i = <49152913>; + }; diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000..338e2a13c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex@perex.cz> + - Takashi Iwai <tiwai@suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + }; + }; |