Squashed 'dts/upstream/' content from commit aaba2d45dc2a
git-subtree-dir: dts/upstream
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diff --git a/Bindings/sound/ac97-bus.txt b/Bindings/sound/ac97-bus.txt
new file mode 100644
index 0000000..103c428
--- /dev/null
+++ b/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/Bindings/sound/adi,adau1372.yaml b/Bindings/sound/adi,adau1372.yaml
new file mode 100644
index 0000000..ea62e51
--- /dev/null
+++ b/Bindings/sound/adi,adau1372.yaml
@@ -0,0 +1,69 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau1372.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+
+title: Analog Devices ADAU1372 CODEC
+
+maintainers:
+ - Alexandre Belloni <alexandre.belloni@bootlin.com>
+
+description: |
+ Analog Devices ADAU1372 four inputs and two outputs codec.
+ https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - adi,adau1372
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: mclk
+
+ powerdown-gpios:
+ description: GPIO used for hardware power-down.
+ maxItems: 1
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@3c {
+ compatible = "adi,adau1372";
+ reg = <0x3c>;
+ #sound-dai-cells = <0>;
+ clock-names = "mclk";
+ clocks = <&adau1372z_xtal>;
+ };
+ };
+
+ adau1372z_xtal: clock {
+ compatible = "fixed-clock";
+ #clock-cells = <0>;
+ clock-frequency = <12288000>;
+ };
+...
diff --git a/Bindings/sound/adi,adau1701.txt b/Bindings/sound/adi,adau1701.txt
new file mode 100644
index 0000000..0d1128c
--- /dev/null
+++ b/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/Bindings/sound/adi,adau17x1.yaml b/Bindings/sound/adi,adau17x1.yaml
new file mode 100644
index 0000000..8ef1e7f
--- /dev/null
+++ b/Bindings/sound/adi,adau17x1.yaml
@@ -0,0 +1,52 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau17x1.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781 Codec
+
+maintainers:
+ - Lars-Peter Clausen <lars@metafoo.de>
+
+properties:
+ compatible:
+ enum:
+ - adi,adau1361
+ - adi,adau1381
+ - adi,adau1461
+ - adi,adau1761
+ - adi,adau1781
+ - adi,adau1961
+
+ reg:
+ maxItems: 1
+ description:
+ The i2c address. Value depends on the state of ADDR0 and ADDR1,
+ as wired in hardware.
+
+ clock-names:
+ const: mclk
+
+ clocks:
+ items:
+ - description: provides the audio master clock for the device.
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@38 {
+ compatible = "adi,adau1761";
+ reg = <0x38>;
+ clock-names = "mclk";
+ clocks = <&audio_clock>;
+ };
+ };
diff --git a/Bindings/sound/adi,adau1977.yaml b/Bindings/sound/adi,adau1977.yaml
new file mode 100644
index 0000000..dba3023
--- /dev/null
+++ b/Bindings/sound/adi,adau1977.yaml
@@ -0,0 +1,94 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau1977.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices ADAU1977/ADAU1978/ADAU1979 Quad ADC with Diagnostics
+
+maintainers:
+ - Lars-Peter Clausen <lars@metafoo.de>
+ - Bogdan Togorean <bogdan.togorean@analog.com>
+
+description: |
+ Analog Devices ADAU1977 and similar quad ADC with Diagnostics
+ https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
+ https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
+ https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
+
+properties:
+ compatible:
+ enum:
+ - adi,adau1977
+ - adi,adau1978
+ - adi,adau1979
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ reset-gpios:
+ maxItems: 1
+
+ AVDD-supply:
+ description: Analog power support for the device.
+
+ DVDD-supply:
+ description: Supply voltage for digital core.
+
+ adi,micbias:
+ description: |
+ Configures the voltage setting for the MICBIAS pin.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1, 2, 3, 4, 5, 6, 7, 8]
+ default: 7
+
+required:
+ - reg
+ - compatible
+ - AVDD-supply
+
+allOf:
+ - $ref: dai-common.yaml#
+ - $ref: /schemas/spi/spi-peripheral-props.yaml#
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ adau1977_spi: adau1977@0 {
+ compatible = "adi,adau1977";
+ reg = <0>;
+ spi-max-frequency = <600000>;
+
+ AVDD-supply = <®ulator>;
+ DVDD-supply = <®ulator_digital>;
+
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+
+ adi,micbias = <3>;
+ };
+ };
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ adau1977_i2c: adau1977@11 {
+ compatible = "adi,adau1977";
+ reg = <0x11>;
+
+ AVDD-supply = <®ulator>;
+ DVDD-supply = <®ulator_digital>;
+
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+ };
+ };
diff --git a/Bindings/sound/adi,adau7002.yaml b/Bindings/sound/adi,adau7002.yaml
new file mode 100644
index 0000000..fcca0fd
--- /dev/null
+++ b/Bindings/sound/adi,adau7002.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: adi,adau7002
+
+ IOVDD-supply:
+ description:
+ IOVDD power supply, if skipped then it is assumed that the supply pin is
+ hardwired to always on.
+
+ wakeup-delay-ms:
+ description:
+ Delay after power up needed for device to settle.
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ audio-codec {
+ compatible = "adi,adau7002";
+ IOVDD-supply = <&pp1800_l15a>;
+ #sound-dai-cells = <0>;
+ wakeup-delay-ms = <80>;
+ };
diff --git a/Bindings/sound/adi,adau7118.yaml b/Bindings/sound/adi,adau7118.yaml
new file mode 100644
index 0000000..12f6050
--- /dev/null
+++ b/Bindings/sound/adi,adau7118.yaml
@@ -0,0 +1,88 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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
+
+unevaluatedProperties: 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/Bindings/sound/adi,axi-i2s.txt b/Bindings/sound/adi,axi-i2s.txt
new file mode 100644
index 0000000..229ad13
--- /dev/null
+++ b/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/Bindings/sound/adi,axi-spdif-tx.txt b/Bindings/sound/adi,axi-spdif-tx.txt
new file mode 100644
index 0000000..7b664e7
--- /dev/null
+++ b/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/Bindings/sound/adi,max98363.yaml b/Bindings/sound/adi,max98363.yaml
new file mode 100644
index 0000000..a844b63
--- /dev/null
+++ b/Bindings/sound/adi,max98363.yaml
@@ -0,0 +1,60 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,max98363.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices MAX98363 SoundWire Amplifier
+
+maintainers:
+ - Ryan Lee <ryans.lee@analog.com>
+
+description:
+ The MAX98363 is a SoundWire input Class D mono amplifier that
+ supports MIPI SoundWire v1.2-compatible digital interface for
+ audio and control data.
+ SoundWire peripheral device ID of MAX98363 is 0x3*019f836300
+ where * is the peripheral device unique ID decoded from pin.
+ It supports up to 10 peripheral devices(0x0 to 0x9).
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: sdw3019f836300
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ soundwire-controller@3250000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x3250000 0x2000>;
+
+ speaker@0,0 {
+ compatible = "sdw3019f836300";
+ reg = <0 0>;
+ #sound-dai-cells = <0>;
+ sound-name-prefix = "Speaker Left";
+ };
+
+ speaker@0,1 {
+ compatible = "sdw3019f836300";
+ reg = <0 1>;
+ #sound-dai-cells = <0>;
+ sound-name-prefix = "Speaker Right";
+ };
+ };
diff --git a/Bindings/sound/adi,max98388.yaml b/Bindings/sound/adi,max98388.yaml
new file mode 100644
index 0000000..93ccd59
--- /dev/null
+++ b/Bindings/sound/adi,max98388.yaml
@@ -0,0 +1,79 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,max98388.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices MAX98388 Speaker Amplifier
+
+maintainers:
+ - Ryan Lee <ryans.lee@analog.com>
+
+description:
+ The MAX98388 is a mono Class-D speaker amplifier with I/V feedback.
+ The device provides a PCM interface for audio data and a standard
+ I2C interface for control data communication.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - adi,max98388
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ adi,vmon-slot-no:
+ description: slot number of the voltage feedback monitor
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 15
+ default: 0
+
+ adi,imon-slot-no:
+ description: slot number of the current feedback monitor
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 15
+ default: 1
+
+ adi,interleave-mode:
+ description:
+ For cases where a single combined channel for the I/V feedback data
+ is not sufficient, the device can also be configured to share
+ a single data output channel on alternating frames.
+ In this configuration, the current and voltage data will be frame
+ interleaved on a single output channel.
+ type: boolean
+
+ reset-gpios:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ max98388: amplifier@39 {
+ compatible = "adi,max98388";
+ reg = <0x39>;
+ #sound-dai-cells = <0>;
+ adi,vmon-slot-no = <0>;
+ adi,imon-slot-no = <1>;
+ adi,interleave-mode;
+ reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>;
+ };
+ };
diff --git a/Bindings/sound/adi,max98396.yaml b/Bindings/sound/adi,max98396.yaml
new file mode 100644
index 0000000..bdc10d4
--- /dev/null
+++ b/Bindings/sound/adi,max98396.yaml
@@ -0,0 +1,141 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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/Bindings/sound/adi,ssm2305.txt b/Bindings/sound/adi,ssm2305.txt
new file mode 100644
index 0000000..a9c9d83
--- /dev/null
+++ b/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/Bindings/sound/adi,ssm2518.yaml b/Bindings/sound/adi,ssm2518.yaml
new file mode 100644
index 0000000..f3f3254
--- /dev/null
+++ b/Bindings/sound/adi,ssm2518.yaml
@@ -0,0 +1,47 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,ssm2518.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices SSM2518 audio amplifier
+
+maintainers:
+ - Lars-Peter Clausen <lars@metafoo.de>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: adi,ssm2518
+
+ reg:
+ maxItems: 1
+ description: |
+ I2C address of the device. This will either be 0x34 (ADDR pin low)
+ or 0x35 (ADDR pin high)
+
+ gpios:
+ maxItems: 1
+ description: |
+ GPIO connected to the nSD pin. If the property is not present
+ it is assumed that the nSD pin is hardwired to always on.
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@34 {
+ compatible = "adi,ssm2518";
+ reg = <0x34>;
+ gpios = <&gpio 5 0>;
+ };
+ };
diff --git a/Bindings/sound/adi,ssm2602.txt b/Bindings/sound/adi,ssm2602.txt
new file mode 100644
index 0000000..3b3302f
--- /dev/null
+++ b/Bindings/sound/adi,ssm2602.txt
@@ -0,0 +1,19 @@
+Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices
+
+SSM2602 support both I2C and SPI as the configuration interface,
+the selection is made by the MODE strap-in pin.
+SSM2603 and SSM2604 only support I2C as the configuration interface.
+
+Required properties:
+
+ - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+ Example:
+
+ ssm2602: ssm2602@1a {
+ compatible = "adi,ssm2602";
+ reg = <0x1a>;
+ };
diff --git a/Bindings/sound/adi,ssm3515.yaml b/Bindings/sound/adi,ssm3515.yaml
new file mode 100644
index 0000000..144450d
--- /dev/null
+++ b/Bindings/sound/adi,ssm3515.yaml
@@ -0,0 +1,49 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,ssm3515.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices SSM3515 Audio Amplifier
+
+maintainers:
+ - Martin Povišer <povik+lin@cutebit.org>
+
+description: |
+ SSM3515 is a mono Class-D audio amplifier with digital input.
+
+ https://www.analog.com/media/en/technical-documentation/data-sheets/SSM3515.pdf
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - adi,ssm3515
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@14 {
+ compatible = "adi,ssm3515";
+ reg = <0x14>;
+ #sound-dai-cells = <0>;
+ sound-name-prefix = "Left Tweeter";
+ };
+ };
diff --git a/Bindings/sound/ak4104.txt b/Bindings/sound/ak4104.txt
new file mode 100644
index 0000000..ae5f7f0
--- /dev/null
+++ b/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/Bindings/sound/ak4118.txt b/Bindings/sound/ak4118.txt
new file mode 100644
index 0000000..6e11a2f
--- /dev/null
+++ b/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/Bindings/sound/ak4375.yaml b/Bindings/sound/ak4375.yaml
new file mode 100644
index 0000000..587598e
--- /dev/null
+++ b/Bindings/sound/ak4375.yaml
@@ -0,0 +1,60 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ak4375.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK4375 DAC and headphones amplifier
+
+maintainers:
+ - Vincent Knecht <vincent.knecht@mailoo.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: asahi-kasei,ak4375
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ avdd-supply:
+ description: regulator phandle for the AVDD power supply.
+
+ tvdd-supply:
+ description: regulator phandle for the TVDD power supply.
+
+ pdn-gpios:
+ description: optional GPIO to set the PDN pin.
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+ - avdd-supply
+ - tvdd-supply
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ headphones: audio-codec@10 {
+ compatible = "asahi-kasei,ak4375";
+ reg = <0x10>;
+ avdd-supply = <®_headphones_avdd>;
+ tvdd-supply = <&pm8916_l6>;
+ pdn-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&headphones_pdn_default>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/ak4554.txt b/Bindings/sound/ak4554.txt
new file mode 100644
index 0000000..934fa02
--- /dev/null
+++ b/Bindings/sound/ak4554.txt
@@ -0,0 +1,11 @@
+AK4554 ADC/DAC
+
+Required properties:
+
+ - compatible : "asahi-kasei,ak4554"
+
+Example:
+
+ak4554-adc-dac {
+ compatible = "asahi-kasei,ak4554";
+};
diff --git a/Bindings/sound/ak4613.yaml b/Bindings/sound/ak4613.yaml
new file mode 100644
index 0000000..75e1341
--- /dev/null
+++ b/Bindings/sound/ak4613.yaml
@@ -0,0 +1,59 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ak4613.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK4613 I2C transmitter
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: asahi-kasei,ak4613
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ ports:
+ $ref: audio-graph-port.yaml#/definitions/ports
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+patternProperties:
+ "^asahi-kasei,in[1-2]-single-end$":
+ description: Input Pin 1 - 2.
+ $ref: /schemas/types.yaml#/definitions/flag
+
+ "^asahi-kasei,out[1-6]-single-end$":
+ description: Output Pin 1 - 6.
+ $ref: /schemas/types.yaml#/definitions/flag
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ ak4613: codec@10 {
+ compatible = "asahi-kasei,ak4613";
+ reg = <0x10>;
+ };
+ };
diff --git a/Bindings/sound/ak4642.yaml b/Bindings/sound/ak4642.yaml
new file mode 100644
index 0000000..437fe5d
--- /dev/null
+++ b/Bindings/sound/ak4642.yaml
@@ -0,0 +1,59 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ak4642.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK4642 I2C transmitter
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - asahi-kasei,ak4642
+ - asahi-kasei,ak4643
+ - asahi-kasei,ak4648
+
+ reg:
+ maxItems: 1
+
+ "#clock-cells":
+ const: 0
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ maxItems: 1
+
+ clock-frequency:
+ description: common clock binding; frequency of MCKO
+
+ clock-output-names:
+ description: common clock name
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ ak4643: codec@12 {
+ compatible = "asahi-kasei,ak4643";
+ #sound-dai-cells = <0>;
+ reg = <0x12>;
+ #clock-cells = <0>;
+ clocks = <&audio_clock>;
+ clock-frequency = <12288000>;
+ clock-output-names = "ak4643_mcko";
+ };
+ };
diff --git a/Bindings/sound/ak5386.txt b/Bindings/sound/ak5386.txt
new file mode 100644
index 0000000..ec3df3a
--- /dev/null
+++ b/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/Bindings/sound/alc5623.txt b/Bindings/sound/alc5623.txt
new file mode 100644
index 0000000..26c86c9
--- /dev/null
+++ b/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/Bindings/sound/allwinner,sun4i-a10-codec.yaml b/Bindings/sound/allwinner,sun4i-a10-codec.yaml
new file mode 100644
index 0000000..7827364
--- /dev/null
+++ b/Bindings/sound/allwinner,sun4i-a10-codec.yaml
@@ -0,0 +1,268 @@
+# 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:
+ - $ref: dai-common.yaml#
+ - 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
+
+unevaluatedProperties: 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/Bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Bindings/sound/allwinner,sun4i-a10-i2s.yaml
new file mode 100644
index 0000000..739114f
--- /dev/null
+++ b/Bindings/sound/allwinner,sun4i-a10-i2s.yaml
@@ -0,0 +1,147 @@
+# 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:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - allwinner,sun6i-a31-i2s
+ - allwinner,sun8i-a83t-i2s
+ - allwinner,sun8i-h3-i2s
+ - allwinner,sun50i-a64-codec-i2s
+ - allwinner,sun50i-h6-i2s
+ - allwinner,sun50i-r329-i2s
+
+ then:
+ required:
+ - resets
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - allwinner,sun8i-a83t-i2s
+ - allwinner,sun8i-h3-i2s
+
+ then:
+ properties:
+ dmas:
+ minItems: 1
+ items:
+ - description: RX DMA Channel
+ - description: TX DMA Channel
+ description:
+ Some controllers cannot receive but can only transmit
+ data. In such a case, the RX DMA channel is to be omitted.
+
+ dma-names:
+ oneOf:
+ - items:
+ - const: rx
+ - const: tx
+ - const: tx
+ description:
+ Some controllers cannot receive but can only transmit
+ data. In such a case, the RX name is to be omitted.
+
+ else:
+ properties:
+ dmas:
+ items:
+ - description: RX DMA Channel
+ - description: TX DMA Channel
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2s0: i2s@1c22400 {
+ #sound-dai-cells = <0>;
+ compatible = "allwinner,sun4i-a10-i2s";
+ reg = <0x01c22400 0x400>;
+ interrupts = <0 16 4>;
+ clocks = <&apb0_gates 3>, <&i2s0_clk>;
+ clock-names = "apb", "mod";
+ dmas = <&dma 0 3>, <&dma 0 3>;
+ dma-names = "rx", "tx";
+ };
+
+...
diff --git a/Bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Bindings/sound/allwinner,sun4i-a10-spdif.yaml
new file mode 100644
index 0000000..8108c56
--- /dev/null
+++ b/Bindings/sound/allwinner,sun4i-a10-spdif.yaml
@@ -0,0 +1,123 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - allwinner,sun6i-a31-spdif
+ - allwinner,sun8i-h3-spdif
+
+ then:
+ required:
+ - resets
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - allwinner,sun8i-h3-spdif
+ - allwinner,sun50i-h6-spdif
+
+ then:
+ properties:
+ dmas:
+ description: TX DMA Channel
+
+ dma-names:
+ const: tx
+
+ else:
+ properties:
+ dmas:
+ items:
+ - description: RX DMA Channel
+ - description: TX DMA Channel
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ spdif: spdif@1c21000 {
+ #sound-dai-cells = <0>;
+ compatible = "allwinner,sun4i-a10-spdif";
+ reg = <0x01c21000 0x40>;
+ interrupts = <13>;
+ clocks = <&apb0_gates 1>, <&spdif_clk>;
+ clock-names = "apb", "spdif";
+ dmas = <&dma 0 2>, <&dma 0 2>;
+ dma-names = "rx", "tx";
+ };
+
+...
diff --git a/Bindings/sound/allwinner,sun50i-a64-codec-analog.yaml b/Bindings/sound/allwinner,sun50i-a64-codec-analog.yaml
new file mode 100644
index 0000000..5800de6
--- /dev/null
+++ b/Bindings/sound/allwinner,sun50i-a64-codec-analog.yaml
@@ -0,0 +1,44 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun50i-a64-codec-analog.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner A64 Analog Codec
+
+maintainers:
+ - Chen-Yu Tsai <wens@csie.org>
+ - Maxime Ripard <mripard@kernel.org>
+
+properties:
+ compatible:
+ const: allwinner,sun50i-a64-codec-analog
+
+ reg:
+ maxItems: 1
+
+ cpvdd-supply:
+ description:
+ Regulator for the headphone amplifier
+
+ allwinner,internal-bias-resistor:
+ description:
+ Enable the internal 2.2K bias resistor between HBIAS and MICDET pins
+ type: boolean
+
+required:
+ - compatible
+ - reg
+ - cpvdd-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ codec_analog: codec-analog@1f015c0 {
+ compatible = "allwinner,sun50i-a64-codec-analog";
+ reg = <0x01f015c0 0x4>;
+ cpvdd-supply = <®_eldo1>;
+ };
+
+...
diff --git a/Bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Bindings/sound/allwinner,sun50i-h6-dmic.yaml
new file mode 100644
index 0000000..763b876
--- /dev/null
+++ b/Bindings/sound/allwinner,sun50i-h6-dmic.yaml
@@ -0,0 +1,87 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner H6 DMIC
+
+maintainers:
+ - Ban Tao <fengzheng923@gmail.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - allwinner,sun20i-d1-dmic
+ - const: allwinner,sun50i-h6-dmic
+ - const: allwinner,sun50i-h6-dmic
+
+ "#sound-dai-cells":
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Bus Clock
+ - description: Module Clock
+
+ clock-names:
+ items:
+ - const: bus
+ - const: mod
+
+ dmas:
+ items:
+ - description: RX DMA Channel
+
+ dma-names:
+ items:
+ - const: rx
+
+ resets:
+ maxItems: 1
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - resets
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ #include <dt-bindings/clock/sun50i-h6-ccu.h>
+ #include <dt-bindings/reset/sun50i-h6-ccu.h>
+
+ dmic: dmic@5095000 {
+ #sound-dai-cells = <0>;
+ compatible = "allwinner,sun50i-h6-dmic";
+ reg = <0x05095000 0x400>;
+ interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>;
+ clock-names = "bus", "mod";
+ dmas = <&dma 7>;
+ dma-names = "rx";
+ resets = <&ccu RST_BUS_DMIC>;
+ };
+
+...
diff --git a/Bindings/sound/allwinner,sun8i-a23-codec-analog.yaml b/Bindings/sound/allwinner,sun8i-a23-codec-analog.yaml
new file mode 100644
index 0000000..1c21a1b
--- /dev/null
+++ b/Bindings/sound/allwinner,sun8i-a23-codec-analog.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a23-codec-analog.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner A23 Analog Codec
+
+maintainers:
+ - Chen-Yu Tsai <wens@csie.org>
+ - Maxime Ripard <mripard@kernel.org>
+
+properties:
+ compatible:
+ oneOf:
+ # FIXME: This is documented in the PRCM binding, but needs to be
+ # migrated here at some point
+ # - allwinner,sun8i-a23-codec-analog
+ - const: allwinner,sun8i-h3-codec-analog
+ - items:
+ - const: allwinner,sun8i-v3-codec-analog
+ - const: allwinner,sun8i-h3-codec-analog
+ - const: allwinner,sun8i-v3s-codec-analog
+
+ reg:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ codec_analog: codec-analog@1f015c0 {
+ compatible = "allwinner,sun8i-h3-codec-analog";
+ reg = <0x01f015c0 0x4>;
+ };
+
+...
diff --git a/Bindings/sound/allwinner,sun8i-a33-codec.yaml b/Bindings/sound/allwinner,sun8i-a33-codec.yaml
new file mode 100644
index 0000000..63eadc4
--- /dev/null
+++ b/Bindings/sound/allwinner,sun8i-a33-codec.yaml
@@ -0,0 +1,68 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ "#sound-dai-cells":
+ minimum: 0
+ maximum: 1
+ description:
+ A value of 0 is deprecated. When used, it only allows access to
+ the ADC/DAC and AIF1 (the CPU DAI), not the other two AIFs/DAIs.
+
+ compatible:
+ oneOf:
+ - items:
+ - const: allwinner,sun50i-a64-codec
+ - const: allwinner,sun8i-a33-codec
+ - const: allwinner,sun8i-a33-codec
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Bus Clock
+ - description: Module Clock
+
+ clock-names:
+ items:
+ - const: bus
+ - const: mod
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ audio-codec@1c22e00 {
+ #sound-dai-cells = <1>;
+ compatible = "allwinner,sun8i-a33-codec";
+ reg = <0x01c22e00 0x400>;
+ interrupts = <0 29 4>;
+ clocks = <&ccu 47>, <&ccu 92>;
+ clock-names = "bus", "mod";
+ };
+
+...
diff --git a/Bindings/sound/amlogic,aiu.yaml b/Bindings/sound/amlogic,aiu.yaml
new file mode 100644
index 0000000..6350dfc
--- /dev/null
+++ b/Bindings/sound/amlogic,aiu.yaml
@@ -0,0 +1,118 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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: dai-common.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/Bindings/sound/amlogic,axg-fifo.yaml b/Bindings/sound/amlogic,axg-fifo.yaml
new file mode 100644
index 0000000..b1b48d6
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-fifo.yaml
@@ -0,0 +1,112 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-fifo.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic AXG Audio FIFO controllers
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ compatible:
+ oneOf:
+ - enum:
+ - amlogic,axg-toddr
+ - amlogic,axg-frddr
+ - items:
+ - enum:
+ - amlogic,g12a-toddr
+ - amlogic,sm1-toddr
+ - const: amlogic,axg-toddr
+ - items:
+ - enum:
+ - amlogic,g12a-frddr
+ - amlogic,sm1-frddr
+ - const: amlogic,axg-frddr
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ items:
+ - description: Peripheral clock
+
+ interrupts:
+ maxItems: 1
+
+ resets:
+ minItems: 1
+ maxItems: 2
+
+ reset-names:
+ minItems: 1
+ maxItems: 2
+
+ amlogic,fifo-depth:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: Size of the controller's fifo in bytes
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - clocks
+ - interrupts
+ - resets
+ - amlogic,fifo-depth
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - amlogic,g12a-toddr
+ - amlogic,sm1-toddr
+ - amlogic,g12a-frddr
+ - amlogic,sm1-frddr
+
+ then:
+ properties:
+ resets:
+ minItems: 2
+ reset-names:
+ items:
+ - const: arb
+ - const: rst
+ required:
+ - reset-names
+
+ else:
+ properties:
+ resets:
+ maxItems: 1
+ reset-names:
+ const: arb
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/axg-audio-clkc.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/reset/amlogic,meson-axg-audio-arb.h>
+ #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h>
+
+ audio-controller@1c0 {
+ compatible = "amlogic,g12a-frddr", "amlogic,axg-frddr";
+ reg = <0x1c0 0x1c>;
+ #sound-dai-cells = <0>;
+ clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
+ interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
+ resets = <&arb>, <&clkc_audio AUD_RESET_FRDDR_A>;
+ reset-names = "arb", "rst";
+ amlogic,fifo-depth = <512>;
+ };
diff --git a/Bindings/sound/amlogic,axg-pdm.yaml b/Bindings/sound/amlogic,axg-pdm.yaml
new file mode 100644
index 0000000..df21dd7
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-pdm.yaml
@@ -0,0 +1,82 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-pdm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG PDM input
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - amlogic,g12a-pdm
+ - amlogic,sm1-pdm
+ - const: amlogic,axg-pdm
+ - const: amlogic,axg-pdm
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ items:
+ - description: Peripheral clock
+ - description: PDM digital clock
+ - description: DSP system clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: dclk
+ - const: sysclk
+
+ resets:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - clocks
+ - clock-names
+
+allOf:
+ - $ref: dai-common.yaml#
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - amlogic,g12a-pdm
+ - amlogic,sm1-pdm
+ then:
+ required:
+ - resets
+
+ else:
+ properties:
+ resets: false
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/axg-audio-clkc.h>
+
+ audio-controller@ff632000 {
+ compatible = "amlogic,axg-pdm";
+ reg = <0xff632000 0x34>;
+ #sound-dai-cells = <0>;
+ clocks = <&clkc_audio AUD_CLKID_PDM>,
+ <&clkc_audio AUD_CLKID_PDM_DCLK>,
+ <&clkc_audio AUD_CLKID_PDM_SYSCLK>;
+ clock-names = "pclk", "dclk", "sysclk";
+ };
diff --git a/Bindings/sound/amlogic,axg-sound-card.yaml b/Bindings/sound/amlogic,axg-sound-card.yaml
new file mode 100644
index 0000000..5db718e
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-sound-card.yaml
@@ -0,0 +1,174 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic AXG sound card
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ const: amlogic,axg-sound-card
+
+ audio-aux-devs:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ description: list of auxiliary devices
+
+ audio-widgets:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description:
+ A list off component DAPM widget. Each entry is a pair of strings,
+ the first being the widget type, the second being the widget name
+
+patternProperties:
+ "^dai-link-[0-9]+$":
+ type: object
+ additionalProperties: false
+ description:
+ Container for dai-link level properties and the CODEC sub-nodes.
+ There should be at least one (and probably more) subnode of this type
+
+ properties:
+ dai-format:
+ $ref: /schemas/types.yaml#/definitions/string
+ enum: [ i2s, left-j, dsp_a ]
+
+ dai-tdm-slot-num:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Number of slots in use. If omitted, slot number is set to
+ accommodate the largest mask provided.
+ maximum: 32
+
+ dai-tdm-slot-width:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: Width in bits for each slot
+ enum: [ 8, 16, 20, 24, 32 ]
+ default: 32
+
+ mclk-fs:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Multiplication factor between the frame rate and master clock
+ rate
+
+ sound-dai:
+ maxItems: 1
+ description: phandle of the CPU DAI
+
+ patternProperties:
+ "^dai-tdm-slot-(t|r)x-mask-[0-3]$":
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 1
+ maxItems: 32
+ description:
+ Transmit and receive cpu slot masks of each TDM lane
+ When omitted, mask is assumed to have to no slots. A valid
+ interface must have at least one slot, so at least one these
+ mask should be provided with an enabled slot.
+
+ "^codec(-[0-9]+)?$":
+ type: object
+ additionalProperties: false
+ description:
+ dai-link representing backend links should have at least one subnode.
+ One subnode for each codec of the dai-link. dai-link representing
+ frontend links have no codec, therefore have no subnodes
+
+ properties:
+ sound-dai:
+ maxItems: 1
+ description: phandle of the codec DAI
+
+ patternProperties:
+ "^dai-tdm-slot-(t|r)x-mask$":
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 1
+ maxItems: 32
+ description: Transmit and receive codec slot masks
+
+ required:
+ - sound-dai
+
+ required:
+ - sound-dai
+
+required:
+ - dai-link-0
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "amlogic,axg-sound-card";
+ model = "AXG-S420";
+ audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
+ audio-widgets = "Line", "Lineout",
+ "Line", "Linein",
+ "Speaker", "Speaker1 Left",
+ "Speaker", "Speaker1 Right",
+ "Speaker", "Speaker2 Left",
+ "Speaker", "Speaker2 Right";
+ audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
+ "SPDIFOUT IN 0", "FRDDR_A OUT 3",
+ "TDM_C Playback", "TDMOUT_C OUT",
+ "TDMIN_A IN 2", "TDM_C Capture",
+ "TDMIN_A IN 5", "TDM_C Loopback",
+ "TODDR_A IN 0", "TDMIN_A OUT",
+ "Lineout", "Lineout AOUTL",
+ "Lineout", "Lineout AOUTR",
+ "Speaker1 Left", "SPK1 OUT_A",
+ "Speaker2 Left", "SPK2 OUT_A",
+ "Speaker1 Right", "SPK1 OUT_B",
+ "Speaker2 Right", "SPK2 OUT_B",
+ "Linein AINL", "Linein",
+ "Linein AINR", "Linein";
+
+ dai-link-0 {
+ sound-dai = <&frddr_a>;
+ };
+
+ dai-link-1 {
+ sound-dai = <&toddr_a>;
+ };
+
+ dai-link-2 {
+ sound-dai = <&tdmif_c>;
+ dai-format = "i2s";
+ dai-tdm-slot-tx-mask-2 = <1 1>;
+ dai-tdm-slot-tx-mask-3 = <1 1>;
+ dai-tdm-slot-rx-mask-1 = <1 1>;
+ mclk-fs = <256>;
+
+ codec-0 {
+ sound-dai = <&lineout>;
+ };
+
+ codec-1 {
+ sound-dai = <&speaker_amp1>;
+ };
+
+ codec-2 {
+ sound-dai = <&speaker_amp2>;
+ };
+
+ codec-3 {
+ sound-dai = <&linein>;
+ };
+ };
+
+ dai-link-3 {
+ sound-dai = <&spdifout>;
+
+ codec {
+ sound-dai = <&spdif_dit>;
+ };
+ };
+ };
diff --git a/Bindings/sound/amlogic,axg-spdifin.yaml b/Bindings/sound/amlogic,axg-spdifin.yaml
new file mode 100644
index 0000000..a0bd7a5
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-spdifin.yaml
@@ -0,0 +1,86 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifin.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG SPDIF Input
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ compatible:
+ oneOf:
+ - const: amlogic,axg-spdifin
+ - items:
+ - enum:
+ - amlogic,g12a-spdifin
+ - amlogic,sm1-spdifin
+ - const: amlogic,axg-spdifin
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ items:
+ - description: Peripheral clock
+ - description: SPDIF input reference clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: refclk
+
+ interrupts:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - clocks
+ - clock-names
+ - interrupts
+
+allOf:
+ - $ref: dai-common.yaml#
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - amlogic,g12a-spdifin
+ - amlogic,sm1-spdifin
+ then:
+ required:
+ - resets
+
+ else:
+ properties:
+ resets: false
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/axg-audio-clkc.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+ audio-controller@400 {
+ compatible = "amlogic,axg-spdifin";
+ reg = <0x400 0x30>;
+ #sound-dai-cells = <0>;
+ interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>;
+ clocks = <&clkc_audio AUD_CLKID_SPDIFIN>,
+ <&clkc_audio AUD_CLKID_SPDIFIN_CLK>;
+ clock-names = "pclk", "refclk";
+ };
diff --git a/Bindings/sound/amlogic,axg-spdifout.yaml b/Bindings/sound/amlogic,axg-spdifout.yaml
new file mode 100644
index 0000000..15be8da
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-spdifout.yaml
@@ -0,0 +1,79 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifout.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG SPDIF Output
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ compatible:
+ oneOf:
+ - const: amlogic,axg-spdifout
+ - items:
+ - enum:
+ - amlogic,g12a-spdifout
+ - amlogic,sm1-spdifout
+ - const: amlogic,axg-spdifout
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ items:
+ - description: Peripheral clock
+ - description: SPDIF output master clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: mclk
+
+ resets:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - clocks
+ - clock-names
+
+allOf:
+ - $ref: dai-common.yaml#
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - amlogic,g12a-spdifout
+ - amlogic,sm1-spdifout
+ then:
+ required:
+ - resets
+
+ else:
+ properties:
+ resets: false
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/axg-audio-clkc.h>
+
+ audio-controller@480 {
+ compatible = "amlogic,axg-spdifout";
+ reg = <0x480 0x50>;
+ #sound-dai-cells = <0>;
+ clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
+ <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
+ clock-names = "pclk", "mclk";
+ };
diff --git a/Bindings/sound/amlogic,axg-tdm-formatters.yaml b/Bindings/sound/amlogic,axg-tdm-formatters.yaml
new file mode 100644
index 0000000..719ca8f
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-tdm-formatters.yaml
@@ -0,0 +1,88 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-formatters.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG TDM formatters
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ compatible:
+ enum:
+ - amlogic,g12a-tdmout
+ - amlogic,sm1-tdmout
+ - amlogic,axg-tdmout
+ - amlogic,g12a-tdmin
+ - amlogic,sm1-tdmin
+ - amlogic,axg-tdmin
+
+ clocks:
+ items:
+ - description: Peripheral clock
+ - description: Bit clock
+ - description: Bit clock input multiplexer
+ - description: Sample clock
+ - description: Sample clock input multiplexer
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: sclk
+ - const: sclk_sel
+ - const: lrclk
+ - const: lrclk_sel
+
+ reg:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+
+allOf:
+ - $ref: component-common.yaml#
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - amlogic,g12a-tdmin
+ - amlogic,sm1-tdmin
+ - amlogic,g12a-tdmout
+ - amlogic,sm1-tdmout
+ then:
+ required:
+ - resets
+
+ else:
+ properties:
+ resets: false
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/axg-audio-clkc.h>
+ #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h>
+
+ audio-controller@500 {
+ compatible = "amlogic,g12a-tdmout";
+ reg = <0x500 0x40>;
+ resets = <&clkc_audio AUD_RESET_TDMOUT_A>;
+ clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
+ <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
+ <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
+ <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
+ <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
+ clock-names = "pclk", "sclk", "sclk_sel",
+ "lrclk", "lrclk_sel";
+ };
diff --git a/Bindings/sound/amlogic,axg-tdm-iface.yaml b/Bindings/sound/amlogic,axg-tdm-iface.yaml
new file mode 100644
index 0000000..45955d8
--- /dev/null
+++ b/Bindings/sound/amlogic,axg-tdm-iface.yaml
@@ -0,0 +1,55 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-iface.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio TDM Interfaces
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: amlogic,axg-tdm-iface
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ minItems: 2
+ items:
+ - description: Bit clock
+ - description: Sample clock
+ - description: Master clock # optional
+
+ clock-names:
+ minItems: 2
+ items:
+ - const: sclk
+ - const: lrclk
+ - const: mclk
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+ - clocks
+ - clock-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/axg-audio-clkc.h>
+
+ audio-controller {
+ compatible = "amlogic,axg-tdm-iface";
+ #sound-dai-cells = <0>;
+ clocks = <&clkc_audio AUD_CLKID_MST_A_SCLK>,
+ <&clkc_audio AUD_CLKID_MST_A_LRCLK>,
+ <&clkc_audio AUD_CLKID_MST_A_MCLK>;
+ clock-names = "sclk", "lrclk", "mclk";
+ };
diff --git a/Bindings/sound/amlogic,g12a-toacodec.yaml b/Bindings/sound/amlogic,g12a-toacodec.yaml
new file mode 100644
index 0000000..23f82bb
--- /dev/null
+++ b/Bindings/sound/amlogic,g12a-toacodec.yaml
@@ -0,0 +1,56 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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: dai-common.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/Bindings/sound/amlogic,g12a-tohdmitx.txt b/Bindings/sound/amlogic,g12a-tohdmitx.txt
new file mode 100644
index 0000000..4e8cd7e
--- /dev/null
+++ b/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/Bindings/sound/amlogic,gx-sound-card.yaml b/Bindings/sound/amlogic,gx-sound-card.yaml
new file mode 100644
index 0000000..d4277d3
--- /dev/null
+++ b/Bindings/sound/amlogic,gx-sound-card.yaml
@@ -0,0 +1,107 @@
+# 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>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ items:
+ - const: amlogic,gx-sound-card
+
+ audio-aux-devs:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ description: list of auxiliary devices
+
+ audio-widgets:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ minItems: 2
+ description: |-
+ A list off component DAPM widget. Each entry is a pair of strings,
+ the first being the widget type, the second being the widget name
+
+patternProperties:
+ "^dai-link-[0-9]+$":
+ 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
+
+unevaluatedProperties: 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/Bindings/sound/amlogic,t9015.yaml b/Bindings/sound/amlogic,t9015.yaml
new file mode 100644
index 0000000..5f5cccd
--- /dev/null
+++ b/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: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^audio-controller@.*"
+
+ "#sound-dai-cells":
+ const: 0
+
+ compatible:
+ items:
+ - const: amlogic,t9015
+
+ clocks:
+ items:
+ - description: Peripheral clock
+
+ clock-names:
+ items:
+ - const: pclk
+
+ reg:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ AVDD-supply:
+ description:
+ Analogue power supply.
+
+ sound-name-prefix: true
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - resets
+ - AVDD-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/g12a-clkc.h>
+ #include <dt-bindings/reset/amlogic,meson-g12a-reset.h>
+
+ acodec: audio-controller@32000 {
+ compatible = "amlogic,t9015";
+ reg = <0x32000 0x14>;
+ #sound-dai-cells = <0>;
+ clocks = <&clkc CLKID_AUDIO_CODEC>;
+ clock-names = "pclk";
+ resets = <&reset RESET_AUDIO_CODEC>;
+ AVDD-supply = <&vddao_1v8>;
+ };
diff --git a/Bindings/sound/apple,mca.yaml b/Bindings/sound/apple,mca.yaml
new file mode 100644
index 0000000..5c6ec08
--- /dev/null
+++ b/Bindings/sound/apple,mca.yaml
@@ -0,0 +1,135 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/apple,mca.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Apple MCA I2S transceiver
+
+description: |
+ MCA is an I2S transceiver peripheral found on M1 and other Apple chips. It is
+ composed of a number of identical clusters which can operate independently
+ or in an interlinked fashion. Up to 6 clusters have been seen on an MCA.
+
+maintainers:
+ - Martin Povišer <povik+lin@cutebit.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ items:
+ - enum:
+ - apple,t6000-mca
+ - apple,t8103-mca
+ - apple,t8112-mca
+ - const: apple,mca
+
+ reg:
+ items:
+ - description: Register region of the MCA clusters proper
+ - description: Register region of the DMA glue and its FIFOs
+
+ interrupts:
+ minItems: 4
+ maxItems: 6
+ description:
+ One interrupt per each cluster
+
+ '#address-cells':
+ const: 1
+
+ '#size-cells':
+ const: 0
+
+ dmas:
+ minItems: 16
+ maxItems: 24
+ description:
+ DMA channels corresponding to the SERDES units in the peripheral. They are
+ listed in groups of four per cluster, and within the group they are given
+ as associated to the TXA, RXA, TXB, RXB units.
+
+ dma-names:
+ minItems: 16
+ items:
+ - const: tx0a
+ - const: rx0a
+ - const: tx0b
+ - const: rx0b
+ - const: tx1a
+ - const: rx1a
+ - const: tx1b
+ - const: rx1b
+ - const: tx2a
+ - const: rx2a
+ - const: tx2b
+ - const: rx2b
+ - const: tx3a
+ - const: rx3a
+ - const: tx3b
+ - const: rx3b
+ - const: tx4a
+ - const: rx4a
+ - const: tx4b
+ - const: rx4b
+ - const: tx5a
+ - const: rx5a
+ - const: tx5b
+ - const: rx5b
+ description: |
+ Names for the DMA channels: 'tx'/'rx', then cluster number, then 'a'/'b'
+ based on the associated SERDES unit.
+
+ clocks:
+ minItems: 4
+ maxItems: 6
+ description:
+ Clusters' input reference clock.
+
+ resets:
+ maxItems: 1
+
+ power-domains:
+ minItems: 5
+ maxItems: 7
+ description:
+ First a general power domain for register access, then the power
+ domains of individual clusters for their operation.
+
+ '#sound-dai-cells':
+ const: 1
+
+required:
+ - compatible
+ - reg
+ - dmas
+ - dma-names
+ - clocks
+ - power-domains
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ mca: i2s@9b600000 {
+ compatible = "apple,t6000-mca", "apple,mca";
+ reg = <0x9b600000 0x10000>,
+ <0x9b200000 0x20000>;
+
+ clocks = <&nco 0>, <&nco 1>, <&nco 2>, <&nco 3>;
+ power-domains = <&ps_audio_p>, <&ps_mca0>, <&ps_mca1>,
+ <&ps_mca2>, <&ps_mca3>;
+ dmas = <&admac 0>, <&admac 1>, <&admac 2>, <&admac 3>,
+ <&admac 4>, <&admac 5>, <&admac 6>, <&admac 7>,
+ <&admac 8>, <&admac 9>, <&admac 10>, <&admac 11>,
+ <&admac 12>, <&admac 13>, <&admac 14>, <&admac 15>;
+ dma-names = "tx0a", "rx0a", "tx0b", "rx0b",
+ "tx1a", "rx1a", "tx1b", "rx1b",
+ "tx2a", "rx2a", "tx2b", "rx2b",
+ "tx3a", "rx3a", "tx3b", "rx3b";
+
+ #sound-dai-cells = <1>;
+ };
diff --git a/Bindings/sound/arm,pl041.yaml b/Bindings/sound/arm,pl041.yaml
new file mode 100644
index 0000000..7896b81
--- /dev/null
+++ b/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/Bindings/sound/armada-370db-audio.txt b/Bindings/sound/armada-370db-audio.txt
new file mode 100644
index 0000000..953c092
--- /dev/null
+++ b/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/Bindings/sound/asahi-kasei,ak4458.yaml b/Bindings/sound/asahi-kasei,ak4458.yaml
new file mode 100644
index 0000000..4477f84
--- /dev/null
+++ b/Bindings/sound/asahi-kasei,ak4458.yaml
@@ -0,0 +1,73 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4458.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK4458 audio DAC
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+ compatible:
+ enum:
+ - asahi-kasei,ak4458
+ - asahi-kasei,ak4497
+
+ reg:
+ maxItems: 1
+
+ avdd-supply:
+ description: Analog power supply
+
+ dvdd-supply:
+ description: Digital power supply
+
+ reset-gpios:
+ maxItems: 1
+
+ mute-gpios:
+ maxItems: 1
+ description:
+ GPIO used to mute all the outputs
+
+ dsd-path:
+ description: Select DSD input pins for ak4497
+ $ref: /schemas/types.yaml#/definitions/uint32
+ oneOf:
+ - const: 0
+ description: "select #16, #17, #19 pins"
+ - const: 1
+ description: "select #3, #4, #5 pins"
+
+required:
+ - compatible
+ - reg
+
+allOf:
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: asahi-kasei,ak4458
+
+ then:
+ properties:
+ dsd-path: false
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@10 {
+ compatible = "asahi-kasei,ak4458";
+ reg = <0x10>;
+ reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>;
+ mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH>;
+ };
+ };
diff --git a/Bindings/sound/asahi-kasei,ak5558.yaml b/Bindings/sound/asahi-kasei,ak5558.yaml
new file mode 100644
index 0000000..d3d494a
--- /dev/null
+++ b/Bindings/sound/asahi-kasei,ak5558.yaml
@@ -0,0 +1,48 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/asahi-kasei,ak5558.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK5558 8 channel differential 32-bit delta-sigma ADC
+
+maintainers:
+ - Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp>
+ - Mihai Serban <mihai.serban@nxp.com>
+
+properties:
+ compatible:
+ enum:
+ - asahi-kasei,ak5552
+ - asahi-kasei,ak5558
+
+ reg:
+ maxItems: 1
+
+ avdd-supply:
+ description: A 1.8V supply that powers up the AVDD pin.
+
+ dvdd-supply:
+ description: A 1.2V supply that powers up the DVDD pin.
+
+ reset-gpios:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ ak5558: codec@10 {
+ compatible = "asahi-kasei,ak5558";
+ reg = <0x10>;
+ reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>;
+ };
+ };
diff --git a/Bindings/sound/atmel,sama5d2-classd.yaml b/Bindings/sound/atmel,sama5d2-classd.yaml
new file mode 100644
index 0000000..43d0470
--- /dev/null
+++ b/Bindings/sound/atmel,sama5d2-classd.yaml
@@ -0,0 +1,100 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 5
+ - 10
+ - 15
+ - 20
+ default: 10
+ description:
+ Set non-overlapping time, the unit is nanosecond(ns).
+ Non-overlapping will be disabled if not specified.
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - dmas
+ - dma-names
+ - clock-names
+ - clocks
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/dma/at91.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+ classd: sound@fc048000 {
+ compatible = "atmel,sama5d2-classd";
+ reg = <0xfc048000 0x100>;
+ interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>;
+ dmas = <&dma0
+ (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
+ | AT91_XDMAC_DT_PERID(47))>;
+ dma-names = "tx";
+ clocks = <&classd_clk>, <&classd_gclk>;
+ clock-names = "pclk", "gclk";
+ assigned-clocks = <&classd_gclk>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_classd_default>;
+ atmel,model = "classd @ SAMA5D2-Xplained";
+ atmel,pwm-type = "diff";
+ atmel,non-overlap-time = <10>;
+ };
diff --git a/Bindings/sound/atmel,sama5d2-i2s.yaml b/Bindings/sound/atmel,sama5d2-i2s.yaml
new file mode 100644
index 0000000..0cd1ff8
--- /dev/null
+++ b/Bindings/sound/atmel,sama5d2-i2s.yaml
@@ -0,0 +1,85 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries
+%YAML 1.2
+---
+$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/Bindings/sound/atmel,sama5d2-pdmic.yaml b/Bindings/sound/atmel,sama5d2-pdmic.yaml
new file mode 100644
index 0000000..f320b56
--- /dev/null
+++ b/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/Bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/Bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt
new file mode 100644
index 0000000..9c5a994
--- /dev/null
+++ b/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/Bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Bindings/sound/atmel-sam9x5-wm8731-audio.txt
new file mode 100644
index 0000000..8facbce
--- /dev/null
+++ b/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/Bindings/sound/atmel-wm8904.txt b/Bindings/sound/atmel-wm8904.txt
new file mode 100644
index 0000000..8bbe50c
--- /dev/null
+++ b/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/Bindings/sound/atmel_ac97c.txt b/Bindings/sound/atmel_ac97c.txt
new file mode 100644
index 0000000..b151bd9
--- /dev/null
+++ b/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/Bindings/sound/audio-graph-card.yaml b/Bindings/sound/audio-graph-card.yaml
new file mode 100644
index 0000000..274092e
--- /dev/null
+++ b/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:
+ - $ref: /schemas/sound/audio-graph.yaml#
+
+properties:
+ compatible:
+ enum:
+ - audio-graph-card
+ - audio-graph-scu-card
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "audio-graph-card";
+
+ dais = <&cpu_port_a>;
+ };
+
+ cpu {
+ /*
+ * dai-controller own settings
+ */
+
+ port {
+ cpu_endpoint: endpoint {
+ remote-endpoint = <&codec_endpoint>;
+ dai-format = "left_j";
+ };
+ };
+ };
+
+ codec {
+ /*
+ * codec own settings
+ */
+
+ port {
+ codec_endpoint: endpoint {
+ remote-endpoint = <&cpu_endpoint>;
+ };
+ };
+ };
diff --git a/Bindings/sound/audio-graph-card2.yaml b/Bindings/sound/audio-graph-card2.yaml
new file mode 100644
index 0000000..d3ce4de
--- /dev/null
+++ b/Bindings/sound/audio-graph-card2.yaml
@@ -0,0 +1,42 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/audio-graph-card2.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Graph Card2
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+ compatible:
+ enum:
+ - audio-graph-card2
+ links:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ label:
+ maxItems: 1
+ routing:
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's source.
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ multi:
+ type: object
+ description: Multi-CPU/Codec node
+ dpcm:
+ type: object
+ description: DPCM node
+ codec2codec:
+ type: object
+ description: Codec to Codec node
+
+required:
+ - compatible
+ - links
+
+additionalProperties: false
+
+...
diff --git a/Bindings/sound/audio-graph-port.yaml b/Bindings/sound/audio-graph-port.yaml
new file mode 100644
index 0000000..60b5e3f
--- /dev/null
+++ b/Bindings/sound/audio-graph-port.yaml
@@ -0,0 +1,116 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/audio-graph-port.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Graph Card 'port'
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+select: false
+
+definitions:
+ port-base:
+ allOf:
+ - $ref: /schemas/graph.yaml#/$defs/port-base
+ - $ref: /schemas/sound/dai-params.yaml#
+ properties:
+ mclk-fs:
+ $ref: simple-card.yaml#/definitions/mclk-fs
+
+ endpoint-base:
+ allOf:
+ - $ref: /schemas/graph.yaml#/$defs/endpoint-base
+ - $ref: /schemas/sound/dai-params.yaml#
+ properties:
+ mclk-fs:
+ $ref: simple-card.yaml#/definitions/mclk-fs
+ frame-inversion:
+ description: dai-link uses frame clock inversion
+ $ref: /schemas/types.yaml#/definitions/flag
+ bitclock-inversion:
+ description: dai-link uses bit clock inversion
+ $ref: /schemas/types.yaml#/definitions/flag
+ frame-master:
+ description: Indicates dai-link frame master.
+ oneOf:
+ - $ref: /schemas/types.yaml#/definitions/flag
+ - $ref: /schemas/types.yaml#/definitions/phandle
+ bitclock-master:
+ description: Indicates dai-link bit clock master
+ oneOf:
+ - $ref: /schemas/types.yaml#/definitions/flag
+ - $ref: /schemas/types.yaml#/definitions/phandle
+ clocks:
+ description: Indicates system clock
+ $ref: /schemas/types.yaml#/definitions/phandle
+ system-clock-frequency:
+ $ref: simple-card.yaml#/definitions/system-clock-frequency
+ system-clock-direction-out:
+ $ref: simple-card.yaml#/definitions/system-clock-direction-out
+ system-clock-fixed:
+ $ref: simple-card.yaml#/definitions/system-clock-fixed
+
+ dai-format:
+ description: audio format.
+ items:
+ enum:
+ - i2s
+ - right_j
+ - left_j
+ - dsp_a
+ - dsp_b
+ - ac97
+ - pdm
+ - msb
+ - lsb
+
+ dai-tdm-slot-num:
+ description: Number of slots in use.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ dai-tdm-slot-width:
+ description: Width in bits for each slot.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ dai-tdm-slot-width-map:
+ description: Mapping of sample widths to slot widths. For hardware
+ that cannot support a fixed slot width or a slot width always
+ equal to sample width. A matrix of one or more 3-tuples.
+ $ref: /schemas/types.yaml#/definitions/uint32-matrix
+ items:
+ items:
+ -
+ description: Sample width in bits
+ minimum: 8
+ maximum: 64
+ -
+ description: Slot width in bits
+ minimum: 8
+ maximum: 256
+ -
+ description: Slot count
+ minimum: 1
+ maximum: 64
+
+ ports:
+ $ref: "#/definitions/port-base"
+ unevaluatedProperties: false
+ patternProperties:
+ "^port(@[0-9a-f]+)?$":
+ $ref: "#/definitions/port-base"
+ unevaluatedProperties: false
+ patternProperties:
+ "^endpoint(@[0-9a-f]+)?":
+ $ref: "#/definitions/endpoint-base"
+ unevaluatedProperties: false
+
+allOf:
+ - $ref: "#/definitions/port-base"
+
+patternProperties:
+ "^endpoint(@[0-9a-f]+)?":
+ $ref: "#/definitions/endpoint-base"
+ unevaluatedProperties: false
+
+additionalProperties: true
diff --git a/Bindings/sound/audio-graph.yaml b/Bindings/sound/audio-graph.yaml
new file mode 100644
index 0000000..71f52f7
--- /dev/null
+++ b/Bindings/sound/audio-graph.yaml
@@ -0,0 +1,47 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/audio-graph.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Graph
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+allOf:
+ - $ref: /schemas/sound/dai-params.yaml#
+
+properties:
+ dais:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ label:
+ maxItems: 1
+ prefix:
+ description: device name prefix
+ $ref: /schemas/types.yaml#/definitions/string
+ routing:
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's source.
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ widgets:
+ description: |
+ User specified audio sound widgets.
+ Each entry is a pair of strings, the first being the type of
+ widget ("Microphone", "Line", "Headphone", "Speaker"), the
+ second being the machine specific name for the widget.
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+
+ pa-gpios:
+ maxItems: 1
+ hp-det-gpio:
+ maxItems: 1
+ mic-det-gpio:
+ maxItems: 1
+
+required:
+ - dais
+
+additionalProperties: true
diff --git a/Bindings/sound/audio-iio-aux.yaml b/Bindings/sound/audio-iio-aux.yaml
new file mode 100644
index 0000000..d3cc1ea
--- /dev/null
+++ b/Bindings/sound/audio-iio-aux.yaml
@@ -0,0 +1,64 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/audio-iio-aux.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio IIO auxiliary
+
+maintainers:
+ - Herve Codina <herve.codina@bootlin.com>
+
+description:
+ Auxiliary device based on Industrial I/O device channels
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: audio-iio-aux
+
+ io-channels:
+ description:
+ Industrial I/O device channels used
+
+ io-channel-names:
+ description:
+ Industrial I/O channel names related to io-channels.
+ These names are used to provides sound controls, widgets and routes names.
+
+ snd-control-invert-range:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description: |
+ A list of 0/1 flags defining whether or not the related channel is
+ inverted
+ items:
+ enum: [0, 1]
+ default: 0
+ description: |
+ Invert the sound control value compared to the IIO channel raw value.
+ - 1: The related sound control value is inverted meaning that the
+ minimum sound control value correspond to the maximum IIO channel
+ raw value and the maximum sound control value correspond to the
+ minimum IIO channel raw value.
+ - 0: The related sound control value is not inverted meaning that the
+ minimum (resp maximum) sound control value correspond to the
+ minimum (resp maximum) IIO channel raw value.
+
+required:
+ - compatible
+ - io-channels
+ - io-channel-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ iio-aux {
+ compatible = "audio-iio-aux";
+ io-channels = <&iio 0>, <&iio 1>, <&iio 2>, <&iio 3>;
+ io-channel-names = "CH0", "CH1", "CH2", "CH3";
+ /* Invert CH1 and CH2 */
+ snd-control-invert-range = <0 1 1 0>;
+ };
diff --git a/Bindings/sound/awinic,aw8738.yaml b/Bindings/sound/awinic,aw8738.yaml
new file mode 100644
index 0000000..bc6c6b1
--- /dev/null
+++ b/Bindings/sound/awinic,aw8738.yaml
@@ -0,0 +1,54 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/awinic,aw8738.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Awinic AW8738 Audio Amplifier
+
+maintainers:
+ - Stephan Gerhold <stephan@gerhold.net>
+
+description:
+ The Awinic AW8738 is a simple audio amplifier with different operation modes
+ (set using one-wire pulse control). The mode configures the speaker-guard
+ function (primarily the power limit for the amplifier).
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: awinic,aw8738
+
+ mode-gpios:
+ description:
+ GPIO used for one-wire pulse control. The pin is typically called SHDN
+ (active-low), but this is misleading since it is actually more than
+ just a simple shutdown/enable control.
+ maxItems: 1
+
+ awinic,mode:
+ description: Operation mode (number of pulses for one-wire pulse control)
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1
+ maximum: 7
+
+ sound-name-prefix: true
+
+required:
+ - compatible
+ - mode-gpios
+ - awinic,mode
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ audio-amplifier {
+ compatible = "awinic,aw8738";
+ mode-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>;
+ awinic,mode = <5>;
+ sound-name-prefix = "Speaker Amp";
+ };
diff --git a/Bindings/sound/awinic,aw87390.yaml b/Bindings/sound/awinic,aw87390.yaml
new file mode 100644
index 0000000..ba9d876
--- /dev/null
+++ b/Bindings/sound/awinic,aw87390.yaml
@@ -0,0 +1,58 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/awinic,aw87390.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Awinic Aw87390 Audio Amplifier
+
+maintainers:
+ - Weidong Wang <wangweidong.a@awinic.com>
+
+description:
+ The awinic aw87390 is specifically designed to improve
+ the musical output dynamic range, enhance the overall
+ sound quallity, which is a new high efficiency, low
+ noise, constant large volume, 6th Smart K audio amplifier.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: awinic,aw87390
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ awinic,audio-channel:
+ description:
+ It is used to distinguish multiple PA devices, so that different
+ configurations can be loaded to different PA devices
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 7
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - awinic,audio-channel
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@58 {
+ compatible = "awinic,aw87390";
+ reg = <0x58>;
+ #sound-dai-cells = <0>;
+ awinic,audio-channel = <0>;
+ };
+ };
diff --git a/Bindings/sound/awinic,aw88395.yaml b/Bindings/sound/awinic,aw88395.yaml
new file mode 100644
index 0000000..ac5f2e0
--- /dev/null
+++ b/Bindings/sound/awinic,aw88395.yaml
@@ -0,0 +1,80 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/awinic,aw88395.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Awinic AW88395 Smart Audio Amplifier
+
+maintainers:
+ - Weidong Wang <wangweidong.a@awinic.com>
+
+description:
+ The Awinic AW88395 is an I2S/TDM input, high efficiency
+ digital Smart K audio amplifier with an integrated 10.25V
+ smart boost convert.
+
+properties:
+ compatible:
+ enum:
+ - awinic,aw88395
+ - awinic,aw88261
+ - awinic,aw88399
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ reset-gpios:
+ maxItems: 1
+
+ awinic,audio-channel:
+ description:
+ It is used to distinguish multiple PA devices, so that different
+ configurations can be loaded to different PA devices
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 7
+
+ awinic,sync-flag:
+ description:
+ Flag bit used to keep the phase synchronized in the case of multiple PA
+ $ref: /schemas/types.yaml#/definitions/flag
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+ - awinic,audio-channel
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - awinic,aw88261
+ then:
+ properties:
+ reset-gpios: false
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@34 {
+ compatible = "awinic,aw88395";
+ reg = <0x34>;
+ #sound-dai-cells = <0>;
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+ awinic,audio-channel = <0>;
+ awinic,sync-flag;
+ };
+ };
diff --git a/Bindings/sound/axentia,tse850-pcm5142.txt b/Bindings/sound/axentia,tse850-pcm5142.txt
new file mode 100644
index 0000000..b6cc5f6
--- /dev/null
+++ b/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 controls two relays, which are either in loop position,
+meaning that input and output are directly connected, or they are in mixer
+position, meaning that the signal is passed through the 'Sum' mixer. Similarly
+for 'loop2'.
+
+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/Bindings/sound/brcm,bcm2835-i2s.txt b/Bindings/sound/brcm,bcm2835-i2s.txt
new file mode 100644
index 0000000..7bb0362
--- /dev/null
+++ b/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/Bindings/sound/brcm,bcm63xx-audio.txt b/Bindings/sound/brcm,bcm63xx-audio.txt
new file mode 100644
index 0000000..007f524
--- /dev/null
+++ b/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/Bindings/sound/brcm,cygnus-audio.txt b/Bindings/sound/brcm,cygnus-audio.txt
new file mode 100644
index 0000000..630bf7c
--- /dev/null
+++ b/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/Bindings/sound/cdns,xtfpga-i2s.txt b/Bindings/sound/cdns,xtfpga-i2s.txt
new file mode 100644
index 0000000..860fc0d
--- /dev/null
+++ b/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/Bindings/sound/cirrus,cs35l41.yaml b/Bindings/sound/cirrus,cs35l41.yaml
new file mode 100644
index 0000000..14dea1f
--- /dev/null
+++ b/Bindings/sound/cirrus,cs35l41.yaml
@@ -0,0 +1,209 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,cs35l41.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Cirrus Logic CS35L41 Speaker Amplifier
+
+maintainers:
+ - david.rhodes@cirrus.com
+
+description: |
+ CS35L41 is a boosted mono Class D amplifier with DSP
+ speaker protection and equalization
+
+properties:
+ compatible:
+ enum:
+ - cirrus,cs35l40
+ - cirrus,cs35l41
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ description:
+ The first cell indicating the audio interface.
+ const: 1
+
+ reset-gpios:
+ maxItems: 1
+
+ VA-supply:
+ description: voltage regulator phandle for the VA supply
+
+ VP-supply:
+ description: voltage regulator phandle for the VP supply
+
+ cirrus,boost-peak-milliamp:
+ description:
+ Boost-converter peak current limit in mA.
+ Configures the peak current by monitoring the current through the boost FET.
+ Range starts at 1600 mA and goes to a maximum of 4500 mA with increments
+ of 50 mA. See section 4.3.6 of the datasheet for details.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1600
+ maximum: 4500
+ default: 4500
+
+ cirrus,boost-ind-nanohenry:
+ description:
+ Boost inductor value, expressed in nH. Valid
+ values include 1000, 1200, 1500 and 2200.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1000
+ maximum: 2200
+
+ cirrus,boost-cap-microfarad:
+ description:
+ Total equivalent boost capacitance on the VBST
+ and VAMP pins, derated at 11 volts DC. The value must be rounded to the
+ nearest integer and expressed in uF.
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ cirrus,asp-sdout-hiz:
+ description:
+ Audio serial port SDOUT Hi-Z control. Sets the Hi-Z
+ configuration for SDOUT pin of amplifier.
+ 0 = Logic 0 during unused slots, and while all transmit channels disabled
+ 1 = Hi-Z during unused slots but logic 0 while all transmit channels disabled
+ 2 = (Default) Logic 0 during unused slots, but Hi-Z while all transmit channels disabled
+ 3 = Hi-Z during unused slots and while all transmit channels disabled
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 3
+ default: 2
+
+ cirrus,boost-type:
+ description:
+ Configures the type of Boost being used.
+ Internal boost requires boost-peak-milliamp, boost-ind-nanohenry and
+ boost-cap-microfarad.
+ External Boost must have GPIO1 as GPIO output. GPIO1 will be set high to
+ enable boost voltage.
+ Shared boost allows two amplifiers to share a single boost circuit by
+ communicating on the MDSYNC bus. The active amplifier controls the boost
+ circuit using combined data from both amplifiers. GPIO1 should be
+ configured for Sync when shared boost is used. Shared boost is not
+ compatible with External boost. Active amplifier requires
+ boost-peak-milliamp, boost-ind-nanohenry and boost-cap-microfarad.
+ 0 = Internal Boost
+ 1 = External Boost
+ 2 = Shared Boost Active
+ 3 = Shared Boost Passive
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 3
+
+ cirrus,gpio1-polarity-invert:
+ description:
+ Boolean which specifies whether the GPIO1
+ level is inverted. If this property is not present the level is not inverted.
+ type: boolean
+
+ cirrus,gpio1-output-enable:
+ description:
+ Boolean which specifies whether the GPIO1 pin
+ is configured as an output. If this property is not present the
+ pin will be configured as an input.
+ type: boolean
+
+ cirrus,gpio1-src-select:
+ description:
+ Configures the function of the GPIO1 pin.
+ Note that the options are different from the GPIO2 pin
+ 0 = High Impedance (Default)
+ 1 = GPIO
+ 2 = Sync
+ 3 = MCLK input
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 3
+
+ cirrus,gpio2-polarity-invert:
+ description:
+ Boolean which specifies whether the GPIO2
+ level is inverted. If this property is not present the level is not inverted.
+ type: boolean
+
+ cirrus,gpio2-output-enable:
+ description:
+ Boolean which specifies whether the GPIO2 pin
+ is configured as an output. If this property is not present the
+ pin will be configured as an input.
+ type: boolean
+
+ cirrus,gpio2-src-select:
+ description:
+ Configures the function of the GPIO2 pin.
+ Note that the options are different from the GPIO1 pin.
+ 0 = High Impedance (Default)
+ 1 = GPIO
+ 2 = Open Drain INTB
+ 3 = MCLK input
+ 4 = Push-pull INTB (active low)
+ 5 = Push-pull INT (active high)
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 5
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ cirrus,boost-type:
+ const: 0
+ then:
+ required:
+ - cirrus,boost-peak-milliamp
+ - cirrus,boost-ind-nanohenry
+ - cirrus,boost-cap-microfarad
+ else:
+ if:
+ properties:
+ cirrus,boost-type:
+ const: 1
+ then:
+ required:
+ - cirrus,gpio1-output-enable
+ - cirrus,gpio1-src-select
+ properties:
+ cirrus,boost-peak-milliamp: false
+ cirrus,boost-ind-nanohenry: false
+ cirrus,boost-cap-microfarad: false
+ cirrus,gpio1-src-select:
+ enum: [1]
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ cs35l41: speaker-amp@2 {
+ #sound-dai-cells = <1>;
+ compatible = "cirrus,cs35l41";
+ reg = <2>;
+ VA-supply = <&dummy_vreg>;
+ VP-supply = <&dummy_vreg>;
+ reset-gpios = <&gpio 110 GPIO_ACTIVE_HIGH>;
+
+ cirrus,boost-type = <0>;
+ cirrus,boost-peak-milliamp = <4500>;
+ cirrus,boost-ind-nanohenry = <1000>;
+ cirrus,boost-cap-microfarad = <15>;
+ };
+ };
diff --git a/Bindings/sound/cirrus,cs35l45.yaml b/Bindings/sound/cirrus,cs35l45.yaml
new file mode 100644
index 0000000..4c9acb8
--- /dev/null
+++ b/Bindings/sound/cirrus,cs35l45.yaml
@@ -0,0 +1,156 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,cs35l45.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Cirrus Logic CS35L45 Speaker Amplifier
+
+maintainers:
+ - Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
+ - Richard Fitzgerald <rf@opensource.cirrus.com>
+
+description: |
+ CS35L45 is a Boosted Mono Class D Amplifier with DSP
+ Speaker Protection and Adaptive Battery Management.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - cirrus,cs35l45
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 1
+
+ reset-gpios:
+ maxItems: 1
+
+ vdd-a-supply:
+ description: voltage regulator phandle for the VDD_A supply
+
+ vdd-batt-supply:
+ description: voltage regulator phandle for the VDD_BATT supply
+
+ spi-max-frequency:
+ maximum: 5000000
+
+ cirrus,asp-sdout-hiz-ctrl:
+ description:
+ Audio serial port SDOUT Hi-Z control. Sets the Hi-Z
+ configuration for SDOUT pin of amplifier. Logical OR of
+ CS35L45_ASP_TX_HIZ_xxx values.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 3
+ default: 2
+
+patternProperties:
+ "^cirrus,gpio-ctrl[1-3]$":
+ description:
+ GPIO pins configuration.
+ type: object
+ additionalProperties: false
+ properties:
+ gpio-dir:
+ description:
+ GPIO pin direction. Valid only when 'gpio-ctrl' is 1
+ 0 = Output
+ 1 = Input
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 1
+ default: 1
+ gpio-lvl:
+ description:
+ GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0
+ 0 = Low
+ 1 = High
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 1
+ default: 0
+ gpio-op-cfg:
+ description:
+ GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0
+ 0 = CMOS
+ 1 = Open Drain
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 1
+ default: 0
+ gpio-pol:
+ description:
+ GPIO output polarity select. Valid only when 'gpio-ctrl' is 1
+ and 'gpio-dir' is 0
+ 0 = Non-inverted, Active High
+ 1 = Inverted, Active Low
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 1
+ default: 0
+ gpio-ctrl:
+ description:
+ Defines the function of the GPIO pin.
+ GPIO1
+ 0 = High impedance input
+ 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir'
+ 2 = Pin acts as MDSYNC, direction controlled by MDSYNC
+ 3-7 = Reserved
+ GPIO2
+ 0 = High impedance input
+ 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir'
+ 2 = Pin acts as open drain INT
+ 3 = Reserved
+ 4 = Pin acts as push-pull output INT. Active low.
+ 5 = Pin acts as push-pull output INT. Active high.
+ 6,7 = Reserved
+ GPIO3
+ 0 = High impedance input
+ 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir'
+ 2-7 = Reserved
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 7
+ default: 0
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/cs35l45.h>
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ cs35l45: cs35l45@2 {
+ #sound-dai-cells = <1>;
+ compatible = "cirrus,cs35l45";
+ reg = <2>;
+ spi-max-frequency = <5000000>;
+ vdd-a-supply = <&dummy_vreg>;
+ vdd-batt-supply = <&dummy_vreg>;
+ reset-gpios = <&gpio 110 0>;
+ cirrus,asp-sdout-hiz-ctrl = <(CS35L45_ASP_TX_HIZ_UNUSED |
+ CS35L45_ASP_TX_HIZ_DISABLED)>;
+ cirrus,gpio-ctrl1 {
+ gpio-ctrl = <0x2>;
+ };
+ cirrus,gpio-ctrl2 {
+ gpio-ctrl = <0x2>;
+ };
+ cirrus,gpio-ctrl3 {
+ gpio-ctrl = <0x1>;
+ gpio-dir = <0x1>;
+ };
+ };
+ };
diff --git a/Bindings/sound/cirrus,cs4234.yaml b/Bindings/sound/cirrus,cs4234.yaml
new file mode 100644
index 0000000..156560b
--- /dev/null
+++ b/Bindings/sound/cirrus,cs4234.yaml
@@ -0,0 +1,74 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,cs4234.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Cirrus Logic cs4234 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+description:
+ The CS4234 is a highly versatile CODEC that combines 4 channels of
+ high performance analog to digital conversion, 4 channels of high
+ performance digital to analog conversion for audio, and 1 channel of
+ digital to analog conversion to provide a nondelayed audio reference
+ signal to an external Class H tracking power supply. If not used to
+ drive a tracking power supply, the 5th DAC can instead be used as a
+ standard audio grade DAC, with performance specifications identical
+ to that of the 4 DACs in the audio path. Additionally, the CS4234
+ includes tunable group delay for each of the 4 audio DAC paths to
+ provide lead time for the external switch-mode power supply, and a
+ nondelayed path into the DAC outputs for input signals requiring a
+ low-latency path to the outputs.
+
+properties:
+ compatible:
+ enum:
+ - cirrus,cs4234
+
+ reg:
+ description:
+ The 7-bit I2C address depends on the state of the ADx pins, in
+ binary given by [0 0 1 0 AD2 AD1 AD0 0].
+ items:
+ minimum: 0x10
+ maximum: 0x17
+
+ VA-supply:
+ description:
+ Analogue power supply.
+
+ VL-supply:
+ description:
+ Interface power supply.
+
+ reset-gpios:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - VA-supply
+ - VL-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c@e0004000 {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ reg = <0xe0004000 0x1000>;
+
+ cs4234: codec@11 {
+ compatible = "cirrus,cs4234";
+ reg = <0x11>;
+
+ VA-supply = <&vdd3v3>;
+ VL-supply = <&vdd3v3>;
+
+ reset-gpios = <&gpio 0>;
+ };
+ };
diff --git a/Bindings/sound/cirrus,cs42l42.yaml b/Bindings/sound/cirrus,cs42l42.yaml
new file mode 100644
index 0000000..af599d8
--- /dev/null
+++ b/Bindings/sound/cirrus,cs42l42.yaml
@@ -0,0 +1,226 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 3
+
+ cirrus,hs-bias-sense-disable:
+ description: |
+ If present the HSBIAS sense is disabled. Configures HSBIAS output
+ current sense through the external 2.21-k resistor. HSBIAS_SENSE
+ is a hardware feature to reduce the potential pop noise when the
+ headset plug is removed slowly. But on some platforms ESD voltage
+ will affect it causing plug detection to fail, especially with CTIA
+ headset type. For different hardware setups, a designer might want
+ to tweak default behavior.
+ type: boolean
+
+required:
+ - compatible
+ - reg
+ - VP-supply
+ - VCP-supply
+ - VD_FILT-supply
+ - VL-supply
+ - VA-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/cs42l42.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ cs42l42: cs42l42@48 {
+ compatible = "cirrus,cs42l42";
+ reg = <0x48>;
+ VA-supply = <&dummy_vreg>;
+ VP-supply = <&dummy_vreg>;
+ VCP-supply = <&dummy_vreg>;
+ VD_FILT-supply = <&dummy_vreg>;
+ VL-supply = <&dummy_vreg>;
+
+ reset-gpios = <&axi_gpio_0 1 0>;
+ interrupt-parent = <&gpio0>;
+ interrupts = <55 8>;
+
+ cirrus,ts-inv = <CS42L42_TS_INV_DIS>;
+ cirrus,ts-dbnc-rise = <CS42L42_TS_DBNCE_1000>;
+ cirrus,ts-dbnc-fall = <CS42L42_TS_DBNCE_0>;
+ cirrus,btn-det-init-dbnce = <100>;
+ cirrus,btn-det-event-dbnce = <10>;
+ cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>;
+ cirrus,hs-bias-ramp-rate = <CS42L42_HSBIAS_RAMP_SLOW>;
+ };
+ };
diff --git a/Bindings/sound/cirrus,cs42l43.yaml b/Bindings/sound/cirrus,cs42l43.yaml
new file mode 100644
index 0000000..7f9d8c7
--- /dev/null
+++ b/Bindings/sound/cirrus,cs42l43.yaml
@@ -0,0 +1,313 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,cs42l43.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Cirrus Logic CS42L43 Audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+description: |
+ The CS42L43 is an audio CODEC with integrated MIPI SoundWire interface
+ (Version 1.2.1 compliant), I2C, SPI, and I2S/TDM interfaces designed
+ for portable applications. It provides a high dynamic range, stereo
+ DAC for headphone output, two integrated Class D amplifiers for
+ loudspeakers, and two ADCs for wired headset microphone input or
+ stereo line input. PDM inputs are provided for digital microphones.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - cirrus,cs42l43
+
+ reg:
+ maxItems: 1
+
+ vdd-p-supply:
+ description:
+ Power supply for the high voltage interface.
+
+ vdd-a-supply:
+ description:
+ Power supply for internal analog circuits.
+
+ vdd-d-supply:
+ description:
+ Power supply for internal digital circuits. Can be internally supplied.
+
+ vdd-io-supply:
+ description:
+ Power supply for external interface and internal digital logic.
+
+ vdd-cp-supply:
+ description:
+ Power supply for the amplifier 3 and 4 charge pump.
+
+ vdd-amp-supply:
+ description:
+ Power supply for amplifier 1 and 2.
+
+ reset-gpios:
+ maxItems: 1
+
+ interrupt-controller: true
+
+ "#interrupt-cells":
+ const: 2
+
+ interrupts:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ clocks:
+ items:
+ - description: Synchronous audio clock provided on mclk_in.
+
+ clock-names:
+ const: mclk
+
+ cirrus,bias-low:
+ type: boolean
+ description:
+ Select a 1.8V headset micbias rather than 2.8V.
+
+ cirrus,bias-sense-microamp:
+ description:
+ Current at which the headset micbias sense clamp will engage, 0 to
+ disable.
+ enum: [ 0, 14, 24, 43, 52, 61, 71, 90, 99 ]
+ default: 14
+
+ cirrus,bias-ramp-ms:
+ description:
+ Time in milliseconds the hardware allows for the headset micbias to
+ ramp up.
+ enum: [ 10, 40, 90, 170 ]
+ default: 170
+
+ cirrus,detect-us:
+ description:
+ Time in microseconds the type detection will run for. Long values will
+ cause more audible effects, but give more accurate detection.
+ enum: [ 20, 100, 1000, 10000, 50000, 75000, 100000, 200000 ]
+ default: 1000
+
+ cirrus,button-automute:
+ type: boolean
+ description:
+ Enable the hardware automuting of decimator 1 when a headset button is
+ pressed.
+
+ cirrus,buttons-ohms:
+ description:
+ Impedance in Ohms for each headset button, these should be listed in
+ ascending order.
+ minItems: 1
+ maxItems: 6
+
+ cirrus,tip-debounce-ms:
+ description:
+ Software debounce on tip sense triggering in milliseconds.
+ default: 0
+
+ cirrus,tip-invert:
+ type: boolean
+ description:
+ Indicates tip detect polarity, inverted implies open-circuit whilst the
+ jack is inserted.
+
+ cirrus,tip-disable-pullup:
+ type: boolean
+ description:
+ Indicates if the internal pullup on the tip detect should be disabled.
+
+ cirrus,tip-fall-db-ms:
+ description:
+ Time in milliseconds a falling edge on the tip detect should be hardware
+ debounced for. Note the falling edge is considered after the invert.
+ enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ]
+ default: 500
+
+ cirrus,tip-rise-db-ms:
+ description:
+ Time in milliseconds a rising edge on the tip detect should be hardware
+ debounced for. Note the rising edge is considered after the invert.
+ enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ]
+ default: 500
+
+ cirrus,use-ring-sense:
+ type: boolean
+ description:
+ Indicates if the ring sense should be used.
+
+ cirrus,ring-invert:
+ type: boolean
+ description:
+ Indicates ring detect polarity, inverted implies open-circuit whilst the
+ jack is inserted.
+
+ cirrus,ring-disable-pullup:
+ type: boolean
+ description:
+ Indicates if the internal pullup on the ring detect should be disabled.
+
+ cirrus,ring-fall-db-ms:
+ description:
+ Time in milliseconds a falling edge on the ring detect should be hardware
+ debounced for. Note the falling edge is considered after the invert.
+ enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ]
+ default: 500
+
+ cirrus,ring-rise-db-ms:
+ description:
+ Time in milliseconds a rising edge on the ring detect should be hardware
+ debounced for. Note the rising edge is considered after the invert.
+ enum: [ 0, 125, 250, 500, 750, 1000, 1250, 1500 ]
+ default: 500
+
+ pinctrl:
+ type: object
+ $ref: /schemas/pinctrl/pinctrl.yaml#
+ additionalProperties: false
+
+ properties:
+ gpio-controller: true
+
+ "#gpio-cells":
+ const: 2
+
+ gpio-ranges:
+ items:
+ - description: A phandle to the CODEC pinctrl node
+ minimum: 0
+ - const: 0
+ - const: 0
+ - const: 3
+
+ patternProperties:
+ "-state$":
+ oneOf:
+ - $ref: "#/$defs/cirrus-cs42l43-state"
+ - patternProperties:
+ "-pins$":
+ $ref: "#/$defs/cirrus-cs42l43-state"
+ additionalProperties: false
+
+ spi:
+ type: object
+ $ref: /schemas/spi/spi-controller.yaml#
+ unevaluatedProperties: false
+
+$defs:
+ cirrus-cs42l43-state:
+ type: object
+
+ allOf:
+ - $ref: /schemas/pinctrl/pincfg-node.yaml#
+ - $ref: /schemas/pinctrl/pinmux-node.yaml#
+
+ oneOf:
+ - required: [ groups ]
+ - required: [ pins ]
+
+ additionalProperties: false
+
+ properties:
+ groups:
+ enum: [ gpio1, gpio2, gpio3, asp, pdmout2, pdmout1, i2c, spi ]
+
+ pins:
+ enum: [ gpio1, gpio2, gpio3,
+ asp_dout, asp_fsync, asp_bclk,
+ pdmout2_clk, pdmout2_data, pdmout1_clk, pdmout1_data,
+ i2c_sda, i2c_scl,
+ spi_miso, spi_sck, spi_ssb ]
+
+ function:
+ enum: [ gpio, spdif, irq, mic-shutter, spk-shutter ]
+
+ drive-strength:
+ description: Set drive strength in mA
+ enum: [ 1, 2, 4, 8, 9, 10, 12, 16 ]
+
+ input-debounce:
+ description: Set input debounce in uS
+ enum: [ 0, 85 ]
+
+required:
+ - compatible
+ - reg
+ - vdd-p-supply
+ - vdd-a-supply
+ - vdd-io-supply
+ - vdd-cp-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ cs42l43: codec@1a {
+ compatible = "cirrus,cs42l43";
+ reg = <0x1a>;
+
+ vdd-p-supply = <&vdd5v0>;
+ vdd-a-supply = <&vdd1v8>;
+ vdd-io-supply = <&vdd1v8>;
+ vdd-cp-supply = <&vdd1v8>;
+ vdd-amp-supply = <&vdd5v0>;
+
+ reset-gpios = <&gpio 0>;
+
+ interrupt-controller;
+ #interrupt-cells = <2>;
+ interrupt-parent = <&gpio>;
+ interrupts = <56 IRQ_TYPE_LEVEL_LOW>;
+
+ #sound-dai-cells = <1>;
+
+ clocks = <&clks 0>;
+ clock-names = "mclk";
+
+ cs42l43_pins: pinctrl {
+ gpio-controller;
+ #gpio-cells = <2>;
+ gpio-ranges = <&cs42l43_pins 0 0 3>;
+
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinsettings>;
+
+ pinsettings: default-state {
+ shutter-pins {
+ groups = "gpio3";
+ function = "mic-shutter";
+ };
+ };
+ };
+
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ cs-gpios = <&cs42l43_pins 1 0>;
+
+ sensor@0 {
+ compatible = "bosch,bme680";
+ reg = <0>;
+ spi-max-frequency = <1400000>;
+ };
+ };
+ };
+ };
diff --git a/Bindings/sound/cirrus,cs42l51.yaml b/Bindings/sound/cirrus,cs42l51.yaml
new file mode 100644
index 0000000..f7bafbd
--- /dev/null
+++ b/Bindings/sound/cirrus,cs42l51.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/cirrus,cs42l51.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: CS42L51 audio codec
+
+maintainers:
+ - Olivier Moysan <olivier.moysan@foss.st.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ cs42l51@4a {
+ compatible = "cirrus,cs42l51";
+ reg = <0x4a>;
+ #sound-dai-cells = <0>;
+ clocks = <&mclk_prov>;
+ clock-names = "MCLK";
+ VL-supply = <®_audio>;
+ VD-supply = <®_audio>;
+ VA-supply = <®_audio>;
+ VAHP-supply = <®_audio>;
+ reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>;
+
+ /* assume audio-graph */
+ port {
+ cpu_endpoint: endpoint {
+ remote-endpoint = <&cpu_endpoint>;
+ };
+ };
+ };
+ };
+...
diff --git a/Bindings/sound/cirrus,ep9301-i2s.yaml b/Bindings/sound/cirrus,ep9301-i2s.yaml
new file mode 100644
index 0000000..453d493
--- /dev/null
+++ b/Bindings/sound/cirrus,ep9301-i2s.yaml
@@ -0,0 +1,66 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,ep9301-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Cirrus EP93xx I2S Controller
+
+description: |
+ The I2S controller is used to stream serial audio data between the external
+ I2S CODECs’, ADCs/DACs, and the ARM Core. The controller supports I2S, Left-
+ and Right-Justified DSP formats.
+
+maintainers:
+ - Alexander Sverdlin <alexander.sverdlin@gmail.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: cirrus,ep9301-i2s
+
+ '#sound-dai-cells':
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ minItems: 3
+ maxItems: 3
+
+ clock-names:
+ items:
+ - const: mclk
+ - const: sclk
+ - const: lrclk
+
+required:
+ - compatible
+ - '#sound-dai-cells'
+ - reg
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |
+ i2s: i2s@80820000 {
+ compatible = "cirrus,ep9301-i2s";
+ #sound-dai-cells = <0>;
+ reg = <0x80820000 0x100>;
+ interrupt-parent = <&vic1>;
+ interrupts = <28>;
+ clocks = <&syscon 29>,
+ <&syscon 30>,
+ <&syscon 31>;
+ clock-names = "mclk", "sclk", "lrclk";
+ };
+
+...
diff --git a/Bindings/sound/cirrus,lochnagar.yaml b/Bindings/sound/cirrus,lochnagar.yaml
new file mode 100644
index 0000000..52f024f
--- /dev/null
+++ b/Bindings/sound/cirrus,lochnagar.yaml
@@ -0,0 +1,55 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - cirrus,lochnagar2-soundcard
+
+ '#sound-dai-cells':
+ description:
+ The first cell indicating the audio interface.
+ const: 1
+
+ clocks:
+ description:
+ Master clock source for the sound card, should normally be set to
+ LOCHNAGAR_SOUNDCARD_MCLK provided by the Lochnagar clock driver.
+ maxItems: 1
+
+ clock-names:
+ const: mclk
+
+required:
+ - compatible
+ - '#sound-dai-cells'
+ - clocks
+ - clock-names
+
+unevaluatedProperties: false
diff --git a/Bindings/sound/cirrus,madera.yaml b/Bindings/sound/cirrus,madera.yaml
new file mode 100644
index 0000000..014d4ea
--- /dev/null
+++ b/Bindings/sound/cirrus,madera.yaml
@@ -0,0 +1,118 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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.
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 2
+ maxItems: 2
+
+additionalProperties: true
diff --git a/Bindings/sound/component-common.yaml b/Bindings/sound/component-common.yaml
new file mode 100644
index 0000000..37766c5
--- /dev/null
+++ b/Bindings/sound/component-common.yaml
@@ -0,0 +1,21 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/component-common.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Component Common Properties
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ sound-name-prefix:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: |
+ Card implementing the routing property define the connection between
+ audio components as list of string pair. Component using the same
+ sink/source names may use this property to prepend the name of their
+ sinks/sources with the provided string.
+
+additionalProperties: true
diff --git a/Bindings/sound/cs35l32.txt b/Bindings/sound/cs35l32.txt
new file mode 100644
index 0000000..1417d3f
--- /dev/null
+++ b/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/Bindings/sound/cs35l33.txt b/Bindings/sound/cs35l33.txt
new file mode 100644
index 0000000..dc5a355
--- /dev/null
+++ b/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/Bindings/sound/cs35l34.txt b/Bindings/sound/cs35l34.txt
new file mode 100644
index 0000000..2f7606b
--- /dev/null
+++ b/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/Bindings/sound/cs35l35.txt b/Bindings/sound/cs35l35.txt
new file mode 100644
index 0000000..e84f30c
--- /dev/null
+++ b/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 Formatting
+ 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/Bindings/sound/cs35l36.txt b/Bindings/sound/cs35l36.txt
new file mode 100644
index 0000000..d34117b
--- /dev/null
+++ b/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 converter 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/Bindings/sound/cs4265.txt b/Bindings/sound/cs4265.txt
new file mode 100644
index 0000000..380fff8
--- /dev/null
+++ b/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/Bindings/sound/cs4270.txt b/Bindings/sound/cs4270.txt
new file mode 100644
index 0000000..c33770e
--- /dev/null
+++ b/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/Bindings/sound/cs4271.txt b/Bindings/sound/cs4271.txt
new file mode 100644
index 0000000..6e699ce
--- /dev/null
+++ b/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/Bindings/sound/cs42l52.txt b/Bindings/sound/cs42l52.txt
new file mode 100644
index 0000000..bc03c93
--- /dev/null
+++ b/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/Bindings/sound/cs42l56.txt b/Bindings/sound/cs42l56.txt
new file mode 100644
index 0000000..4ba520a
--- /dev/null
+++ b/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/Bindings/sound/cs42l73.txt b/Bindings/sound/cs42l73.txt
new file mode 100644
index 0000000..47b868b
--- /dev/null
+++ b/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/Bindings/sound/cs42xx8.txt b/Bindings/sound/cs42xx8.txt
new file mode 100644
index 0000000..bbfe393
--- /dev/null
+++ b/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/Bindings/sound/cs43130.txt b/Bindings/sound/cs43130.txt
new file mode 100644
index 0000000..8b1dd5a
--- /dev/null
+++ b/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/Bindings/sound/cs4341.txt b/Bindings/sound/cs4341.txt
new file mode 100644
index 0000000..12b4aa8
--- /dev/null
+++ b/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/Bindings/sound/cs4349.txt b/Bindings/sound/cs4349.txt
new file mode 100644
index 0000000..54c117b
--- /dev/null
+++ b/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/Bindings/sound/cs53l30.txt b/Bindings/sound/cs53l30.txt
new file mode 100644
index 0000000..dc256ad
--- /dev/null
+++ b/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 single 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/Bindings/sound/da7213.txt b/Bindings/sound/da7213.txt
new file mode 100644
index 0000000..94584c9
--- /dev/null
+++ b/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/Bindings/sound/da7218.txt b/Bindings/sound/da7218.txt
new file mode 100644
index 0000000..2cf3089
--- /dev/null
+++ b/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/Bindings/sound/da9055.txt b/Bindings/sound/da9055.txt
new file mode 100644
index 0000000..75c6338
--- /dev/null
+++ b/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/Bindings/sound/dai-common.yaml b/Bindings/sound/dai-common.yaml
new file mode 100644
index 0000000..1aed2f0
--- /dev/null
+++ b/Bindings/sound/dai-common.yaml
@@ -0,0 +1,18 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/dai-common.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Digital Audio Interface Common Properties
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+allOf:
+ - $ref: component-common.yaml#
+
+properties:
+ '#sound-dai-cells': true
+
+additionalProperties: true
diff --git a/Bindings/sound/dai-params.yaml b/Bindings/sound/dai-params.yaml
new file mode 100644
index 0000000..cd85081
--- /dev/null
+++ b/Bindings/sound/dai-params.yaml
@@ -0,0 +1,37 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/dai-params.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Digital Audio Interface (DAI) Stream Parameters
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+select: false
+
+properties:
+ convert-channels:
+ description: Number of audio channels used by DAI
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1
+ maximum: 32
+
+ convert-sample-format:
+ description: Audio sample format used by DAI
+ $ref: /schemas/types.yaml#/definitions/string
+ enum:
+ - s8
+ - s16_le
+ - s24_le
+ - s24_3le
+ - s32_le
+
+ convert-rate:
+ description: Audio sample rate used by DAI
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 8000
+ maximum: 192000
+
+additionalProperties: true
diff --git a/Bindings/sound/davinci-evm-audio.txt b/Bindings/sound/davinci-evm-audio.txt
new file mode 100644
index 0000000..963e100
--- /dev/null
+++ b/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/Bindings/sound/davinci-mcasp-audio.yaml b/Bindings/sound/davinci-mcasp-audio.yaml
new file mode 100644
index 0000000..7735e08
--- /dev/null
+++ b/Bindings/sound/davinci-mcasp-audio.yaml
@@ -0,0 +1,202 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: McASP Controller for TI SoCs
+
+maintainers:
+ - Jayesh Choudhary <j-choudhary@ti.com>
+
+properties:
+ compatible:
+ enum:
+ - ti,dm646x-mcasp-audio
+ - ti,da830-mcasp-audio
+ - ti,am33xx-mcasp-audio
+ - ti,dra7-mcasp-audio
+ - ti,omap4-mcasp-audio
+
+ reg:
+ minItems: 1
+ items:
+ - description: CFG registers
+ - description: data registers
+
+ reg-names:
+ minItems: 1
+ items:
+ - const: mpu
+ - const: dat
+
+ op-mode:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: 0 - I2S or 1 - DIT operation mode
+ enum:
+ - 0
+ - 1
+
+ tdm-slots:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ number of channels over one serializer
+ the property is ignored in DIT mode
+ minimum: 2
+ maximum: 32
+
+ serial-dir:
+ description:
+ A list of serializer configuration
+ Entry is indication for serializer pin direction
+ 0 - Inactive, 1 - TX, 2 - RX
+ All AXR pins should be present in the array even if inactive
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 1
+ maxItems: 25
+ items:
+ minimum: 0
+ maximum: 2
+
+ dmas:
+ minItems: 1
+ items:
+ - description: transmission DMA channel
+ - description: reception DMA channel
+
+ dma-names:
+ minItems: 1
+ items:
+ - const: tx
+ - const: rx
+
+ ti,hwmods:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: Name of hwmod associated with McASP
+ maxItems: 1
+ deprecated: true
+
+ tx-num-evt:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ configures WFIFO threshold
+ 0 disables the FIFO use
+ if property is missing, then also FIFO use is disabled
+
+ rx-num-evt:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ configures RFIFO threshold
+ 0 disables the FIFO use
+ if property is missing, then also FIFO use is disabled
+
+ dismod:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ specify the drive on TX pin during inactive time slots
+ 0 - 3-state, 2 - logic low, 3 - logic high
+ enum:
+ - 0
+ - 2
+ - 3
+ default: 2
+
+ interrupts:
+ anyOf:
+ - minItems: 1
+ items:
+ - description: TX interrupt
+ - description: RX interrupt
+ - items:
+ - description: common/combined interrupt
+
+ interrupt-names:
+ oneOf:
+ - minItems: 1
+ items:
+ - const: tx
+ - const: rx
+ - const: common
+
+ fck_parent:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: parent clock name for McASP fck
+ maxItems: 1
+
+ auxclk-fs-ratio:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: ratio of AUCLK and FS rate if applicable
+
+ gpio-controller: true
+
+ "#gpio-cells":
+ const: 2
+
+ clocks:
+ minItems: 1
+ items:
+ - description: functional clock
+ - description: module specific optional ahclkx clock
+ - description: module specific optional ahclkr clock
+
+ clock-names:
+ minItems: 1
+ items:
+ - const: fck
+ - const: ahclkx
+ - const: ahclkr
+
+ power-domains:
+ description: phandle to the corresponding power-domain
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ port:
+ description: connection for when McASP is used via graph card
+ type: object
+
+required:
+ - compatible
+ - reg
+ - reg-names
+ - dmas
+ - dma-names
+ - interrupts
+ - interrupt-names
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ opmode:
+ enum:
+ - 0
+
+ then:
+ required:
+ - tdm-slots
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ mcasp0: mcasp0@1d00000 {
+ compatible = "ti,da830-mcasp-audio";
+ reg = <0x100000 0x3000>;
+ reg-names = "mpu";
+ interrupts = <82>, <83>;
+ interrupt-names = "tx", "rx";
+ op-mode = <0>; /* MCASP_IIS_MODE */
+ tdm-slots = <2>;
+ dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>;
+ dma-names = "tx", "rx";
+ serial-dir = <
+ 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */
+ 0 0 0 0
+ 0 0 0 1
+ 2 0 0 0 >;
+ tx-num-evt = <1>;
+ rx-num-evt = <1>;
+ };
diff --git a/Bindings/sound/davinci-mcbsp.txt b/Bindings/sound/davinci-mcbsp.txt
new file mode 100644
index 0000000..3ffc256
--- /dev/null
+++ b/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/Bindings/sound/dialog,da7219.yaml b/Bindings/sound/dialog,da7219.yaml
new file mode 100644
index 0000000..19137ab
--- /dev/null
+++ b/Bindings/sound/dialog,da7219.yaml
@@ -0,0 +1,238 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Dialog Semiconductor DA7219 Audio Codec
+
+maintainers:
+ - David Rau <David.Rau.opensource@dm.renesas.com>
+
+description:
+ The DA7219 is an ultra low-power audio codec with
+ in-built advanced accessory detection (AAD) for mobile
+ computing and accessory applications, which supports
+ sample rates up to 96 kHz at 24-bit resolution.
+
+properties:
+ compatible:
+ const: dlg,da7219
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ VDD-supply:
+ description:
+ VDD power supply for the device.
+
+ VDDMIC-supply:
+ description:
+ VDDMIC power supply for the device.
+
+ VDDIO-supply:
+ description:
+ VDDIO power supply for the device.
+
+ interrupt-names:
+ description:
+ Should be "wakeup" if interrupt is to be used to wake system,
+ otherwise "irq" should be used.
+ enum:
+ - wakeup
+ - irq
+
+ wakeup-source:
+ type: boolean
+ description:
+ Flag to indicate this device can wake system (suspend/resume).
+
+ "#clock-cells":
+ const: 1
+
+ clock-output-names:
+ minItems: 2
+ maxItems: 2
+ description:
+ Name given for DAI WCLK and BCLK outputs.
+
+ clocks:
+ maxItems: 1
+ description:
+ phandle and clock specifier for codec MCLK.
+
+ clock-names:
+ const: mclk
+
+ dlg,micbias-lvl:
+ enum: [1600, 1800, 2000, 2200, 2400, 2600]
+ description:
+ Voltage (mV) for Mic Bias.
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,mic-amp-in-sel:
+ enum: [diff, se_p, se_n]
+ description:
+ Mic input source type.
+
+ diff - Differential.
+
+ se_p - MIC_P.
+ Positive differential analog microphone input.
+
+ se_n - MIC_N.
+ Negative differential analog microphone input.
+ $ref: /schemas/types.yaml#/definitions/string
+
+ da7219_aad:
+ type: object
+ additionalProperties: false
+ description:
+ Configuration of advanced accessory detection.
+ properties:
+ dlg,micbias-pulse-lvl:
+ enum: [2800, 2900]
+ description:
+ Mic bias higher voltage pulse level (mV).
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,micbias-pulse-time:
+ description:
+ Mic bias higher voltage pulse duration (ms).
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+
+ dlg,btn-cfg:
+ enum: [2, 5, 10, 50, 100, 200, 500]
+ description:
+ Periodic button press measurements for 4-pole jack (ms).
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,mic-det-thr:
+ enum: [200, 500, 750, 1000]
+ description:
+ Impedance threshold for mic detection measurement (Ohms).
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,jack-ins-deb:
+ enum: [5, 10, 20, 50, 100, 200, 500, 1000]
+ description:
+ Debounce time for jack insertion (ms).
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,jack-ins-det-pty:
+ enum: [low, high]
+ description:
+ Polarity for jack insertion detection.
+ $ref: /schemas/types.yaml#/definitions/string
+
+ dlg,jack-det-rate:
+ enum: ["32_64", "64_128", "128_256", "256_512"]
+ description:
+ Jack type (3/4 pole) detection latency (ms).
+ $ref: /schemas/types.yaml#/definitions/string
+
+ dlg,jack-rem-deb:
+ enum: [1, 5, 10, 20]
+ description:
+ Debounce time for jack removal (ms).
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,a-d-btn-thr:
+ description:
+ Impedance threshold between buttons A and D.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 255
+
+ dlg,d-b-btn-thr:
+ description:
+ Impedance threshold between buttons D and B.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 255
+
+ dlg,b-c-btn-thr:
+ description:
+ Impedance threshold between buttons B and C.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 255
+
+ dlg,c-mic-btn-thr:
+ description:
+ Impedance threshold between button C and Mic.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 255
+
+ dlg,btn-avg:
+ enum: [1, 2, 4, 8]
+ description:
+ Number of 8-bit readings for averaged button measurement.
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dlg,adc-1bit-rpt:
+ enum: [1, 2, 4, 8]
+ description:
+ Repeat count for 1-bit button measurement.
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - VDD-supply
+ - VDDMIC-supply
+ - VDDIO-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/irq.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec: da7219@1a {
+ compatible = "dlg,da7219";
+ reg = <0x1a>;
+
+ interrupt-parent = <&gpio6>;
+ interrupts = <11 IRQ_TYPE_LEVEL_LOW>;
+
+ VDD-supply = <&vdd_reg>;
+ VDDMIC-supply = <&vddmic_reg>;
+ VDDIO-supply = <&vddio_reg>;
+
+ #clock-cells = <1>;
+ clock-output-names = "da7219-dai-wclk", "da7219-dai-bclk";
+
+ clocks = <&clks 201>;
+ clock-names = "mclk";
+
+ dlg,micbias-lvl = <2600>;
+ dlg,mic-amp-in-sel = "diff";
+
+ da7219_aad {
+ dlg,btn-cfg = <50>;
+ dlg,mic-det-thr = <500>;
+ dlg,jack-ins-deb = <20>;
+ dlg,jack-ins-det-pty = "low";
+ dlg,jack-det-rate = "32_64";
+ dlg,jack-rem-deb = <1>;
+
+ dlg,a-d-btn-thr = <0xa>;
+ dlg,d-b-btn-thr = <0x16>;
+ dlg,b-c-btn-thr = <0x21>;
+ dlg,c-mic-btn-thr = <0x3E>;
+
+ dlg,btn-avg = <4>;
+ dlg,adc-1bit-rpt = <1>;
+ };
+ };
+ };
diff --git a/Bindings/sound/dmic-codec.yaml b/Bindings/sound/dmic-codec.yaml
new file mode 100644
index 0000000..59ef0cf
--- /dev/null
+++ b/Bindings/sound/dmic-codec.yaml
@@ -0,0 +1,55 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/dmic-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Generic PDM Digital microphone (DMIC) codec
+
+maintainers:
+ - Arnaud Pouliquen <arnaud.pouliquen@foss.st.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: dmic-codec
+
+ '#sound-dai-cells':
+ const: 0
+
+ dmicen-gpios:
+ description: GPIO specifier for DMIC to control start and stop
+ maxItems: 1
+
+ num-channels:
+ description: Number of microphones on this DAI
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1
+ maximum: 8
+ default: 8
+
+ modeswitch-delay-ms:
+ description: Delay (in ms) to complete DMIC mode switch
+
+ wakeup-delay-ms:
+ description: Delay (in ms) after enabling the DMIC
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ dmic {
+ compatible = "dmic-codec";
+ dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>;
+ num-channels = <1>;
+ wakeup-delay-ms = <50>;
+ modeswitch-delay-ms = <35>;
+ };
+...
diff --git a/Bindings/sound/es8328.txt b/Bindings/sound/es8328.txt
new file mode 100644
index 0000000..33fbf05
--- /dev/null
+++ b/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/Bindings/sound/eukrea-tlv320.txt b/Bindings/sound/eukrea-tlv320.txt
new file mode 100644
index 0000000..6dfa88c
--- /dev/null
+++ b/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/Bindings/sound/everest,es7134.txt b/Bindings/sound/everest,es7134.txt
new file mode 100644
index 0000000..0916660
--- /dev/null
+++ b/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/Bindings/sound/everest,es7241.txt b/Bindings/sound/everest,es7241.txt
new file mode 100644
index 0000000..28f82cf
--- /dev/null
+++ b/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/Bindings/sound/everest,es8316.yaml b/Bindings/sound/everest,es8316.yaml
new file mode 100644
index 0000000..b6079b3
--- /dev/null
+++ b/Bindings/sound/everest,es8316.yaml
@@ -0,0 +1,57 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/everest,es8316.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Everest ES8316 audio CODEC
+
+maintainers:
+ - Daniel Drake <drake@endlessm.com>
+ - Katsuhiro Suzuki <katsuhiro@katsuster.net>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: everest,es8316
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: clock for master clock (MCLK)
+
+ clock-names:
+ items:
+ - const: mclk
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ es8316: codec@11 {
+ compatible = "everest,es8316";
+ reg = <0x11>;
+ clocks = <&clks 10>;
+ clock-names = "mclk";
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/everest,es8326.yaml b/Bindings/sound/everest,es8326.yaml
new file mode 100644
index 0000000..0778140
--- /dev/null
+++ b/Bindings/sound/everest,es8326.yaml
@@ -0,0 +1,116 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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/Bindings/sound/fsl,asrc.txt b/Bindings/sound/fsl,asrc.txt
new file mode 100644
index 0000000..998b4c8
--- /dev/null
+++ b/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/Bindings/sound/fsl,aud2htx.yaml b/Bindings/sound/fsl,aud2htx.yaml
new file mode 100644
index 0000000..aa4be71
--- /dev/null
+++ b/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/Bindings/sound/fsl,audmix.txt b/Bindings/sound/fsl,audmix.txt
new file mode 100644
index 0000000..840b7e0
--- /dev/null
+++ b/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/Bindings/sound/fsl,easrc.yaml b/Bindings/sound/fsl,easrc.yaml
new file mode 100644
index 0000000..a680d7a
--- /dev/null
+++ b/Bindings/sound/fsl,easrc.yaml
@@ -0,0 +1,106 @@
+# 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:
+ oneOf:
+ - enum:
+ - fsl,imx8mn-easrc
+ - items:
+ - enum:
+ - fsl,imx8mp-easrc
+ - 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/Bindings/sound/fsl,esai.txt b/Bindings/sound/fsl,esai.txt
new file mode 100644
index 0000000..90112ca
--- /dev/null
+++ b/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 Transmission 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/Bindings/sound/fsl,micfil.yaml b/Bindings/sound/fsl,micfil.yaml
new file mode 100644
index 0000000..b7e6058
--- /dev/null
+++ b/Bindings/sound/fsl,micfil.yaml
@@ -0,0 +1,89 @@
+# 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
+ - fsl,imx93-micfil
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ items:
+ - description: Digital Microphone interface interrupt
+ - description: Digital Microphone interface error interrupt
+ - description: voice activity detector event interrupt
+ - description: voice activity detector error interrupt
+
+ dmas:
+ items:
+ - description: DMA controller phandle and request line for RX
+
+ dma-names:
+ items:
+ - const: rx
+
+ clocks:
+ items:
+ - description: The ipg clock for register access
+ - description: internal micfil clock
+ - description: PLL clock source for 8kHz series
+ - description: PLL clock source for 11kHz series
+ - description: External clock 3
+ minItems: 2
+
+ clock-names:
+ items:
+ - const: ipg_clk
+ - const: ipg_clk_app
+ - const: pll8k
+ - const: pll11k
+ - const: clkext3
+ minItems: 2
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - dmas
+ - dma-names
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/imx8mm-clock.h>
+ micfil: audio-controller@30080000 {
+ compatible = "fsl,imx8mm-micfil";
+ reg = <0x30080000 0x10000>;
+ interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 44 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 45 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&clk IMX8MM_CLK_PDM_IPG>,
+ <&clk IMX8MM_CLK_PDM_ROOT>;
+ clock-names = "ipg_clk", "ipg_clk_app";
+ dmas = <&sdma2 24 25 0>;
+ dma-names = "rx";
+ };
diff --git a/Bindings/sound/fsl,mqs.txt b/Bindings/sound/fsl,mqs.txt
new file mode 100644
index 0000000..d66284b
--- /dev/null
+++ b/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/Bindings/sound/fsl,qmc-audio.yaml b/Bindings/sound/fsl,qmc-audio.yaml
new file mode 100644
index 0000000..b522ed7
--- /dev/null
+++ b/Bindings/sound/fsl,qmc-audio.yaml
@@ -0,0 +1,118 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,qmc-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: QMC audio
+
+maintainers:
+ - Herve Codina <herve.codina@bootlin.com>
+
+description: |
+ The QMC audio is an ASoC component which uses QMC (QUICC Multichannel
+ Controller) channels to transfer the audio data.
+ It provides as many DAI as the number of QMC channel used.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: fsl,qmc-audio
+
+ '#address-cells':
+ const: 1
+ '#size-cells':
+ const: 0
+ '#sound-dai-cells':
+ const: 1
+
+patternProperties:
+ '^dai@([0-9]|[1-5][0-9]|6[0-3])$':
+ description:
+ A DAI managed by this controller
+ type: object
+ additionalProperties: false
+
+ properties:
+ reg:
+ minimum: 0
+ maximum: 63
+ description:
+ The DAI number
+
+ fsl,qmc-chan:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ items:
+ - items:
+ - description: phandle to QMC node
+ - description: Channel number
+ description:
+ Should be a phandle/number pair. The phandle to QMC node and the QMC
+ channel to use for this DAI.
+
+ required:
+ - reg
+ - fsl,qmc-chan
+
+required:
+ - compatible
+ - '#address-cells'
+ - '#size-cells'
+ - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+ - |
+ audio_controller: audio-controller {
+ compatible = "fsl,qmc-audio";
+ #address-cells = <1>;
+ #size-cells = <0>;
+ #sound-dai-cells = <1>;
+ dai@16 {
+ reg = <16>;
+ fsl,qmc-chan = <&qmc 16>;
+ };
+ dai@17 {
+ reg = <17>;
+ fsl,qmc-chan = <&qmc 17>;
+ };
+ };
+
+ sound {
+ compatible = "simple-audio-card";
+ #address-cells = <1>;
+ #size-cells = <0>;
+ simple-audio-card,dai-link@0 {
+ reg = <0>;
+ format = "dsp_b";
+ cpu {
+ sound-dai = <&audio_controller 16>;
+ };
+ codec {
+ sound-dai = <&codec1>;
+ dai-tdm-slot-num = <4>;
+ dai-tdm-slot-width = <8>;
+ /* TS 3, 5, 7, 9 */
+ dai-tdm-slot-tx-mask = <0 0 0 1 0 1 0 1 0 1>;
+ dai-tdm-slot-rx-mask = <0 0 0 1 0 1 0 1 0 1>;
+ };
+ };
+ simple-audio-card,dai-link@1 {
+ reg = <1>;
+ format = "dsp_b";
+ cpu {
+ sound-dai = <&audio_controller 17>;
+ };
+ codec {
+ sound-dai = <&codec2>;
+ dai-tdm-slot-num = <4>;
+ dai-tdm-slot-width = <8>;
+ /* TS 2, 4, 6, 8 */
+ dai-tdm-slot-tx-mask = <0 0 1 0 1 0 1 0 1>;
+ dai-tdm-slot-rx-mask = <0 0 1 0 1 0 1 0 1>;
+ };
+ };
+ };
diff --git a/Bindings/sound/fsl,rpmsg.yaml b/Bindings/sound/fsl,rpmsg.yaml
new file mode 100644
index 0000000..188f38b
--- /dev/null
+++ b/Bindings/sound/fsl,rpmsg.yaml
@@ -0,0 +1,133 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio RPMSG CPU DAI Controller
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+description: |
+ fsl_rpmsg is a virtual audio device. Mapping to real hardware devices
+ are SAI, MICFIL, DMA controlled by Cortex M core. What we see from
+ Linux side is a device which provides audio service by rpmsg channel.
+ We can create different sound cards which access different hardwares
+ such as SAI, MICFIL, .etc through building rpmsg channels between
+ Cortex-A and Cortex-M.
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - fsl,imx7ulp-rpmsg-audio
+ - fsl,imx8mn-rpmsg-audio
+ - fsl,imx8mm-rpmsg-audio
+ - fsl,imx8mp-rpmsg-audio
+ - fsl,imx8ulp-rpmsg-audio
+ - fsl,imx93-rpmsg-audio
+
+ clocks:
+ items:
+ - description: Peripheral clock for register access
+ - description: Master clock
+ - description: DMA clock for DMA register access
+ - description: Parent clock for multiple of 8kHz sample rates
+ - description: Parent clock for multiple of 11kHz sample rates
+
+ clock-names:
+ items:
+ - const: ipg
+ - const: mclk
+ - const: dma
+ - const: pll8k
+ - const: pll11k
+
+ power-domains:
+ description:
+ List of phandle and PM domain specifier as documented in
+ Documentation/devicetree/bindings/power/power_domain.txt
+ maxItems: 1
+
+ memory-region:
+ maxItems: 1
+ description:
+ phandle to a node describing reserved memory (System RAM memory)
+ The M core can't access all the DDR memory space on some platform,
+ So reserved a specific memory for dma buffer which M core can
+ access.
+ (see bindings/reserved-memory/reserved-memory.txt)
+
+ audio-codec:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle to a node of audio codec
+
+ fsl,enable-lpa:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: enable low power audio path.
+
+ fsl,rpmsg-out:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: |
+ This is a boolean property. If present, the transmitting function
+ will be enabled.
+
+ fsl,rpmsg-in:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: |
+ This is a boolean property. If present, the receiving function
+ will be enabled.
+
+ fsl,rpmsg-channel-name:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: |
+ A string property to assign rpmsg channel this sound card sits on.
+ This property can be omitted if there is only one sound card and it sits
+ on "rpmsg-audio-channel".
+ enum:
+ - rpmsg-audio-channel
+ - rpmsg-micfil-channel
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/imx8mn-clock.h>
+
+ rpmsg_audio: rpmsg_audio {
+ compatible = "fsl,imx8mn-rpmsg-audio";
+ model = "wm8524-audio";
+ fsl,enable-lpa;
+ fsl,rpmsg-out;
+ clocks = <&clk IMX8MN_CLK_SAI3_IPG>,
+ <&clk IMX8MN_CLK_SAI3_ROOT>,
+ <&clk IMX8MN_CLK_SDMA3_ROOT>,
+ <&clk IMX8MN_AUDIO_PLL1_OUT>,
+ <&clk IMX8MN_AUDIO_PLL2_OUT>;
+ clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k";
+ };
+
+ - |
+ #include <dt-bindings/clock/imx8mm-clock.h>
+
+ rpmsg_micfil: audio-controller {
+ compatible = "fsl,imx8mm-rpmsg-audio";
+ model = "micfil-audio";
+ fsl,rpmsg-channel-name = "rpmsg-micfil-channel";
+ fsl,enable-lpa;
+ fsl,rpmsg-in;
+ clocks = <&clk IMX8MM_CLK_PDM_IPG>,
+ <&clk IMX8MM_CLK_PDM_ROOT>,
+ <&clk IMX8MM_CLK_SDMA3_ROOT>,
+ <&clk IMX8MM_AUDIO_PLL1_OUT>,
+ <&clk IMX8MM_AUDIO_PLL2_OUT>;
+ clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k";
+ };
+
+...
diff --git a/Bindings/sound/fsl,sai.yaml b/Bindings/sound/fsl,sai.yaml
new file mode 100644
index 0000000..088c26b
--- /dev/null
+++ b/Bindings/sound/fsl,sai.yaml
@@ -0,0 +1,203 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,sai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Synchronous Audio Interface (SAI).
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+description: |
+ The SAI is based on I2S module that used communicating with audio codecs,
+ which provides a synchronous audio interface that supports fullduplex
+ serial interfaces with frame synchronization such as I2S, AC97, TDM, and
+ codec/DSP interfaces.
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - fsl,imx6ul-sai
+ - fsl,imx7d-sai
+ - const: fsl,imx6sx-sai
+
+ - items:
+ - enum:
+ - fsl,imx8mm-sai
+ - fsl,imx8mn-sai
+ - fsl,imx8mp-sai
+ - const: fsl,imx8mq-sai
+
+ - items:
+ - enum:
+ - fsl,imx6sx-sai
+ - fsl,imx7ulp-sai
+ - fsl,imx8mq-sai
+ - fsl,imx8qm-sai
+ - fsl,imx8ulp-sai
+ - fsl,imx93-sai
+ - fsl,vf610-sai
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: The ipg clock for register access
+ - description: master clock source 0 (obsoleted)
+ - description: master clock source 1
+ - description: master clock source 2
+ - description: master clock source 3
+ - description: PLL clock source for 8kHz series
+ - description: PLL clock source for 11kHz series
+ minItems: 4
+
+ clock-names:
+ oneOf:
+ - items:
+ - const: bus
+ - const: mclk0
+ - const: mclk1
+ - const: mclk2
+ - const: mclk3
+ - const: pll8k
+ - const: pll11k
+ minItems: 5
+ - items:
+ - const: bus
+ - const: mclk1
+ - const: mclk2
+ - const: mclk3
+ - const: pll8k
+ - const: pll11k
+ minItems: 4
+
+ 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
+
+ interrupts:
+ items:
+ - description: receive and transmit interrupt
+
+ big-endian:
+ description: |
+ required if all the SAI registers are big-endian rather than little-endian.
+ type: boolean
+
+ fsl,dataline:
+ $ref: /schemas/types.yaml#/definitions/uint32-matrix
+ description: |
+ Configure the dataline. It has 3 value for each configuration
+ maxItems: 16
+ items:
+ items:
+ - description: format Default(0), I2S(1) or PDM(2)
+ enum: [0, 1, 2]
+ - description: dataline mask for 'rx'
+ - description: dataline mask for 'tx'
+
+ fsl,sai-mclk-direction-output:
+ description: SAI will output the SAI MCLK clock.
+ type: boolean
+
+ fsl,sai-synchronous-rx:
+ description: |
+ SAI will work in the synchronous mode (sync Tx with Rx) which means
+ both the transmitter and the receiver will send and receive data by
+ following receiver's bit clocks and frame sync clocks.
+ type: boolean
+
+ fsl,sai-asynchronous:
+ description: |
+ SAI will work in the asynchronous mode, which means both transmitter
+ and receiver will send and receive data by following their own bit clocks
+ and frame sync clocks separately.
+ If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
+ default synchronous mode (sync Rx with Tx) will be used, which means both
+ transmitter and receiver will send and receive data by following clocks
+ of transmitter.
+ type: boolean
+
+ fsl,shared-interrupt:
+ description: Interrupt is shared with other modules.
+ type: boolean
+
+ lsb-first:
+ description: |
+ Configures whether the LSB or the MSB is transmitted
+ first for the fifo data. If this property is absent,
+ the MSB is transmitted first as default, or the LSB
+ is transmitted first.
+ type: boolean
+
+ "#sound-dai-cells":
+ const: 0
+ description: optional, some dts node didn't add it.
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ required:
+ - fsl,sai-asynchronous
+ then:
+ properties:
+ fsl,sai-synchronous-rx: false
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - interrupts
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/vf610-clock.h>
+ sai2: sai@40031000 {
+ compatible = "fsl,vf610-sai";
+ reg = <0x40031000 0x1000>;
+ interrupts = <86 IRQ_TYPE_LEVEL_HIGH>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_sai2_1>;
+ clocks = <&clks VF610_CLK_PLATFORM_BUS>,
+ <&clks VF610_CLK_SAI2>,
+ <&clks 0>, <&clks 0>;
+ clock-names = "bus", "mclk1", "mclk2", "mclk3";
+ dma-names = "rx", "tx";
+ dmas = <&edma0 0 20>, <&edma0 0 21>;
+ big-endian;
+ lsb-first;
+ };
+
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/imx8mm-clock.h>
+ sai1: sai@30010000 {
+ compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai";
+ reg = <0x30010000 0x10000>;
+ interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&clk IMX8MM_CLK_SAI1_IPG>,
+ <&clk IMX8MM_CLK_DUMMY>,
+ <&clk IMX8MM_CLK_SAI1_ROOT>,
+ <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>;
+ clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3";
+ dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>;
+ dma-names = "rx", "tx";
+ fsl,dataline = <1 0xff 0xff 2 0xff 0x11>;
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/fsl,spdif.yaml b/Bindings/sound/fsl,spdif.yaml
new file mode 100644
index 0000000..1d64e83
--- /dev/null
+++ b/Bindings/sound/fsl,spdif.yaml
@@ -0,0 +1,120 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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/Bindings/sound/fsl,ssi.txt b/Bindings/sound/fsl,ssi.txt
new file mode 100644
index 0000000..7e15a85
--- /dev/null
+++ b/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/Bindings/sound/fsl,xcvr.yaml b/Bindings/sound/fsl,xcvr.yaml
new file mode 100644
index 0000000..799b362
--- /dev/null
+++ b/Bindings/sound/fsl,xcvr.yaml
@@ -0,0 +1,105 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,xcvr.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio Transceiver (XCVR) Controller
+
+maintainers:
+ - Viorel Suman <viorel.suman@nxp.com>
+
+description: |
+ NXP XCVR (Audio Transceiver) is a on-chip functional module
+ that allows CPU to receive and transmit digital audio via
+ HDMI2.1 eARC, HDMI1.4 ARC and SPDIF.
+
+properties:
+ $nodename:
+ pattern: "^xcvr@.*"
+
+ compatible:
+ enum:
+ - fsl,imx8mp-xcvr
+ - fsl,imx93-xcvr
+
+ reg:
+ items:
+ - description: 20K RAM for code and data
+ - description: registers space
+ - description: RX FIFO address
+ - description: TX FIFO address
+
+ reg-names:
+ items:
+ - const: ram
+ - const: regs
+ - const: rxfifo
+ - const: txfifo
+
+ interrupts:
+ 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/Bindings/sound/fsl-asoc-card.txt b/Bindings/sound/fsl-asoc-card.txt
new file mode 100644
index 0000000..4e8dbc5
--- /dev/null
+++ b/Bindings/sound/fsl-asoc-card.txt
@@ -0,0 +1,117 @@
+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"
+
+ "fsl,imx-audio-nau8822"
+
+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/Bindings/sound/google,chv3-codec.yaml b/Bindings/sound/google,chv3-codec.yaml
new file mode 100644
index 0000000..5329dc1
--- /dev/null
+++ b/Bindings/sound/google,chv3-codec.yaml
@@ -0,0 +1,31 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/google,chv3-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Google Chameleon v3 audio codec
+
+maintainers:
+ - Paweł Anikiel <pan@semihalf.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: google,chv3-codec
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+
+additionalProperties: false
+
+examples:
+ - |
+ audio-codec {
+ compatible = "google,chv3-codec";
+ };
diff --git a/Bindings/sound/google,chv3-i2s.yaml b/Bindings/sound/google,chv3-i2s.yaml
new file mode 100644
index 0000000..3ce910f
--- /dev/null
+++ b/Bindings/sound/google,chv3-i2s.yaml
@@ -0,0 +1,44 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/google,chv3-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Google Chameleon v3 I2S device
+
+maintainers:
+ - Paweł Anikiel <pan@semihalf.com>
+
+description: |
+ I2S device for the Google Chameleon v3. The device handles both RX
+ and TX using a producer/consumer ring buffer design.
+
+properties:
+ compatible:
+ const: google,chv3-i2s
+
+ reg:
+ items:
+ - description: core registers
+ - description: irq registers
+
+ interrupts:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - interrupts
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+ i2s@c0060300 {
+ compatible = "google,chv3-i2s";
+ reg = <0xc0060300 0x100>,
+ <0xc0060f00 0x10>;
+ interrupts = <GIC_SPI 20 IRQ_TYPE_LEVEL_HIGH>;
+ };
diff --git a/Bindings/sound/google,cros-ec-codec.yaml b/Bindings/sound/google,cros-ec-codec.yaml
new file mode 100644
index 0000000..1434f44
--- /dev/null
+++ b/Bindings/sound/google,cros-ec-codec.yaml
@@ -0,0 +1,78 @@
+# SPDX-License-Identifier: GPL-2.0-only
+%YAML 1.2
+---
+$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).
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: google,cros-ec-codec
+
+ "#sound-dai-cells":
+ const: 1
+
+ reg:
+ items:
+ - description: |
+ Physical base address and length of shared memory region from EC.
+ It contains 3 unsigned 32-bit integer. The first 2 integers
+ combine to become an unsigned 64-bit physical address.
+ The last one integer is the length of the shared memory.
+
+ memory-region:
+ maxItems: 1
+ description: |
+ Shared memory region to EC. A "shared-dma-pool".
+ See ../reserved-memory/reserved-memory.txt for details.
+
+required:
+ - compatible
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ reserved_mem: reserved-mem@52800000 {
+ compatible = "shared-dma-pool";
+ reg = <0x52800000 0x100000>;
+ no-map;
+ };
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ cros-ec@0 {
+ compatible = "google,cros-ec-spi";
+ reg = <0>;
+ interrupts = <93 0>;
+
+ codecs {
+ #address-cells = <2>;
+ #size-cells = <1>;
+
+ cros_ec_codec: ec-codec@10500000 {
+ compatible = "google,cros-ec-codec";
+ #sound-dai-cells = <1>;
+ reg = <0x0 0x10500000 0x80000>;
+ memory-region = <&reserved_mem>;
+ };
+
+ };
+ };
+ };
diff --git a/Bindings/sound/google,sc7180-trogdor.yaml b/Bindings/sound/google,sc7180-trogdor.yaml
new file mode 100644
index 0000000..bac9405
--- /dev/null
+++ b/Bindings/sound/google,sc7180-trogdor.yaml
@@ -0,0 +1,137 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/google,sc7180-trogdor.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Google SC7180-Trogdor ASoC sound card driver
+
+maintainers:
+ - Rohit kumar <quic_rohkumar@quicinc.com>
+ - Cheng-Yi Chiang <cychiang@chromium.org>
+
+description:
+ This binding describes the SC7180 sound card which uses LPASS for audio.
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - google,sc7180-trogdor
+ - google,sc7180-coachz
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 0
+
+ dmic-gpios:
+ maxItems: 1
+ description: GPIO for switching between DMICs
+
+patternProperties:
+ "^dai-link(@[0-9])?$":
+ description:
+ Each subnode represents a dai link. Subnodes of each dai links would be
+ cpu/codec dais.
+
+ type: object
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name.
+ $ref: /schemas/types.yaml#/definitions/string
+ maxItems: 1
+
+ reg:
+ maxItems: 1
+ description: dai link address.
+
+ cpu:
+ description: Holds subnode which indicates cpu dai.
+ type: object
+ additionalProperties: false
+
+ properties:
+ sound-dai:
+ maxItems: 1
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 4
+
+ required:
+ - link-name
+ - cpu
+ - codec
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - "#address-cells"
+ - "#size-cells"
+
+unevaluatedProperties: false
+
+examples:
+
+ - |
+ sound {
+ compatible = "google,sc7180-trogdor";
+ model = "sc7180-rt5682-max98357a-2mic";
+
+ audio-routing =
+ "Headphone Jack", "HPOL",
+ "Headphone Jack", "HPOR";
+
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ dmic-gpios = <&tlmm 86 0>;
+
+ dai-link@0 {
+ link-name = "MultiMedia0";
+ reg = <0>;
+ cpu {
+ sound-dai = <&lpass_cpu 0>;
+ };
+
+ codec {
+ sound-dai = <&alc5682 0>;
+ };
+ };
+
+ dai-link@1 {
+ link-name = "MultiMedia1";
+ reg = <1>;
+ cpu {
+ sound-dai = <&lpass_cpu 1>;
+ };
+
+ codec {
+ sound-dai = <&max98357a>;
+ };
+ };
+
+ dai-link@2 {
+ link-name = "MultiMedia2";
+ reg = <2>;
+ cpu {
+ sound-dai = <&lpass_hdmi 0>;
+ };
+
+ codec {
+ sound-dai = <&msm_dp>;
+ };
+ };
+ };
diff --git a/Bindings/sound/google,sc7280-herobrine.yaml b/Bindings/sound/google,sc7280-herobrine.yaml
new file mode 100644
index 0000000..ec4b6e5
--- /dev/null
+++ b/Bindings/sound/google,sc7280-herobrine.yaml
@@ -0,0 +1,183 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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.
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - google,sc7280-herobrine
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 0
+
+patternProperties:
+ "^dai-link@[0-9a-f]$":
+ description:
+ Each subnode represents a dai link. Subnodes of each dai links would be
+ cpu/codec dais.
+
+ type: object
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name.
+ $ref: /schemas/types.yaml#/definitions/string
+ maxItems: 1
+
+ reg:
+ maxItems: 1
+ description: dai link address.
+
+ cpu:
+ description: Holds subnode which indicates cpu dai.
+ type: object
+ properties:
+ sound-dai: true
+
+ required:
+ - sound-dai
+
+ additionalProperties: false
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ properties:
+ sound-dai: true
+
+ required:
+ - sound-dai
+
+ additionalProperties: false
+
+ platform:
+ description: Holds subnode which includes the phandle of q6apm platform device.
+ type: object
+ properties:
+ sound-dai:
+ maxItems: 1
+
+ required:
+ - sound-dai
+
+ additionalProperties: false
+
+ required:
+ - link-name
+ - cpu
+ - codec
+ - reg
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - "#address-cells"
+ - "#size-cells"
+
+unevaluatedProperties: false
+
+examples:
+
+ - |
+ #include <dt-bindings/sound/qcom,lpass.h>
+ sound {
+ compatible = "google,sc7280-herobrine";
+ model = "sc7280-wcd938x-max98360a-4dmic";
+
+ audio-routing =
+ "IN1_HPHL", "HPHL_OUT",
+ "IN2_HPHR", "HPHR_OUT",
+ "AMIC1", "MIC BIAS1",
+ "AMIC2", "MIC BIAS2",
+ "VA DMIC0", "MIC BIAS3",
+ "VA DMIC1", "MIC BIAS3",
+ "VA DMIC2", "MIC BIAS4",
+ "VA DMIC3", "MIC BIAS4",
+ "TX SWR_ADC0", "ADC1_OUTPUT",
+ "TX SWR_ADC1", "ADC2_OUTPUT",
+ "TX SWR_ADC2", "ADC3_OUTPUT",
+ "TX SWR_DMIC0", "DMIC1_OUTPUT",
+ "TX SWR_DMIC1", "DMIC2_OUTPUT",
+ "TX SWR_DMIC2", "DMIC3_OUTPUT",
+ "TX SWR_DMIC3", "DMIC4_OUTPUT";
+
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ dai-link@0 {
+ link-name = "WCD Playback";
+ reg = <LPASS_CDC_DMA_RX0>;
+ cpu {
+ sound-dai = <&lpass_cpu LPASS_CDC_DMA_RX0>;
+ };
+
+ codec {
+ sound-dai = <&wcd938x 0>, <&swr0 0>, <&rxmacro 0>;
+ };
+ };
+ dai-link@1 {
+ link-name = "WCD Capture";
+ reg = <LPASS_CDC_DMA_TX3>;
+ cpu {
+ sound-dai = <&lpass_cpu LPASS_CDC_DMA_TX3>;
+ };
+
+ codec {
+ sound-dai = <&wcd938x 1>, <&swr1 0>, <&txmacro 0>;
+ };
+ };
+
+ dai-link@2 {
+ link-name = "MI2S Playback";
+ reg = <MI2S_SECONDARY>;
+ cpu {
+ sound-dai = <&lpass_cpu MI2S_SECONDARY>;
+ };
+
+ codec {
+ sound-dai = <&max98360a>;
+ };
+ };
+
+ dai-link@3 {
+ link-name = "DMIC Capture";
+ reg = <LPASS_CDC_DMA_VA_TX0>;
+ cpu {
+ sound-dai = <&lpass_cpu LPASS_CDC_DMA_VA_TX0>;
+ };
+
+ codec {
+ sound-dai = <&vamacro 0>;
+ };
+ };
+
+ dai-link@5 {
+ link-name = "DP Playback";
+ reg = <LPASS_DP_RX>;
+ cpu {
+ sound-dai = <&lpass_cpu LPASS_DP_RX>;
+ };
+
+ codec {
+ sound-dai = <&mdss_dp>;
+ };
+ };
+ };
diff --git a/Bindings/sound/hisilicon,hi6210-i2s.txt b/Bindings/sound/hisilicon,hi6210-i2s.txt
new file mode 100644
index 0000000..7a29678
--- /dev/null
+++ b/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/Bindings/sound/ics43432.txt b/Bindings/sound/ics43432.txt
new file mode 100644
index 0000000..e6f05f2
--- /dev/null
+++ b/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/Bindings/sound/img,i2s-in.txt b/Bindings/sound/img,i2s-in.txt
new file mode 100644
index 0000000..423265c
--- /dev/null
+++ b/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/Bindings/sound/img,i2s-out.txt b/Bindings/sound/img,i2s-out.txt
new file mode 100644
index 0000000..6b0ee9b
--- /dev/null
+++ b/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/Bindings/sound/img,parallel-out.txt b/Bindings/sound/img,parallel-out.txt
new file mode 100644
index 0000000..37a3f94c
--- /dev/null
+++ b/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/Bindings/sound/img,pistachio-internal-dac.txt b/Bindings/sound/img,pistachio-internal-dac.txt
new file mode 100644
index 0000000..4cc18fc
--- /dev/null
+++ b/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/Bindings/sound/img,spdif-in.txt b/Bindings/sound/img,spdif-in.txt
new file mode 100644
index 0000000..f7ea8c8
--- /dev/null
+++ b/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/Bindings/sound/img,spdif-out.txt b/Bindings/sound/img,spdif-out.txt
new file mode 100644
index 0000000..413ed8b
--- /dev/null
+++ b/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/Bindings/sound/imx-audio-card.yaml b/Bindings/sound/imx-audio-card.yaml
new file mode 100644
index 0000000..f7ad5ea
--- /dev/null
+++ b/Bindings/sound/imx-audio-card.yaml
@@ -0,0 +1,117 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/imx-audio-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP i.MX audio sound card.
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - fsl,imx-audio-card
+
+patternProperties:
+ ".*-dai-link$":
+ description:
+ Each subnode represents a dai link. Subnodes of each dai links would be
+ cpu/codec dais.
+
+ type: object
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name.
+ $ref: /schemas/types.yaml#/definitions/string
+ maxItems: 1
+
+ format:
+ description: audio format.
+ items:
+ enum:
+ - i2s
+ - dsp_b
+
+ dai-tdm-slot-num:
+ description: see tdm-slot.txt.
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ dai-tdm-slot-width:
+ description: see tdm-slot.txt.
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ cpu:
+ description: Holds subnode which indicates cpu dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ maxItems: 1
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+
+ fsl,mclk-equal-bclk:
+ description: Indicates mclk can be equal to bclk, especially for sai interface
+ $ref: /schemas/types.yaml#/definitions/flag
+
+ required:
+ - link-name
+ - cpu
+
+ additionalProperties: false
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound-ak4458 {
+ compatible = "fsl,imx-audio-card";
+ model = "ak4458-audio";
+ pri-dai-link {
+ link-name = "akcodec";
+ format = "i2s";
+ fsl,mclk-equal-bclk;
+ cpu {
+ sound-dai = <&sai1>;
+ };
+ codec {
+ sound-dai = <&ak4458_1>, <&ak4458_2>;
+ };
+ };
+ fe-dai-link {
+ link-name = "HiFi-ASRC-FE";
+ format = "i2s";
+ cpu {
+ sound-dai = <&easrc>;
+ };
+ };
+ be-dai-link {
+ link-name = "HiFi-ASRC-BE";
+ format = "dsp_b";
+ dai-tdm-slot-num = <8>;
+ dai-tdm-slot-width = <32>;
+ fsl,mclk-equal-bclk;
+ cpu {
+ sound-dai = <&sai1>;
+ };
+ codec {
+ sound-dai = <&ak4458_1>, <&ak4458_2>;
+ };
+ };
+ };
diff --git a/Bindings/sound/imx-audio-es8328.txt b/Bindings/sound/imx-audio-es8328.txt
new file mode 100644
index 0000000..07b68ab
--- /dev/null
+++ b/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/Bindings/sound/imx-audio-hdmi.yaml b/Bindings/sound/imx-audio-hdmi.yaml
new file mode 100644
index 0000000..e7e7bb6
--- /dev/null
+++ b/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/Bindings/sound/imx-audio-sgtl5000.txt b/Bindings/sound/imx-audio-sgtl5000.txt
new file mode 100644
index 0000000..2f89db8
--- /dev/null
+++ b/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/Bindings/sound/imx-audio-spdif.txt b/Bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 0000000..da84a44
--- /dev/null
+++ b/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/Bindings/sound/imx-audmux.yaml b/Bindings/sound/imx-audmux.yaml
new file mode 100644
index 0000000..dab45c3
--- /dev/null
+++ b/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:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description: |
+ List of configuration options for the specific port.
+ For imx31-audmux and above, it is a list of tuples ptcr pdcr.
+ For imx21-audmux it is a list of pcr values.
+
+ required:
+ - fsl,audmux-port
+ - fsl,port-config
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ audmux@21d8000 {
+ compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux";
+ reg = <0x021d8000 0x4000>;
+ };
+ - |
+ audmux@10016000 {
+ compatible = "fsl,imx27-audmux", "fsl,imx21-audmux";
+ reg = <0x10016000 0x1000>;
+ clocks = <&clks 1>;
+ clock-names = "audmux";
+
+ mux-ssi0 {
+ fsl,audmux-port = <0>;
+ fsl,port-config = <0xcb205000>;
+ };
+
+ mux-pins4 {
+ fsl,audmux-port = <2>;
+ fsl,port-config = <0x00001000>;
+ };
+ };
+ - |
+ #include <dt-bindings/sound/fsl-imx-audmux.h>
+ audmux@21d8000 {
+ compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux";
+ reg = <0x021d8000 0x4000>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_audmux>;
+
+ mux-ssi1 {
+ fsl,audmux-port = <0>;
+ fsl,port-config = <
+ IMX_AUDMUX_V2_PTCR_SYN 0
+ IMX_AUDMUX_V2_PTCR_TFSEL(2) 0
+ IMX_AUDMUX_V2_PTCR_TCSEL(2) 0
+ IMX_AUDMUX_V2_PTCR_TFSDIR 0
+ IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2)
+ >;
+ };
+
+ mux-pins3 {
+ fsl,audmux-port = <2>;
+ fsl,port-config = <
+ IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0)
+ 0 IMX_AUDMUX_V2_PDCR_TXRXEN
+ >;
+ };
+ };
diff --git a/Bindings/sound/infineon,peb2466.yaml b/Bindings/sound/infineon,peb2466.yaml
new file mode 100644
index 0000000..66993d3
--- /dev/null
+++ b/Bindings/sound/infineon,peb2466.yaml
@@ -0,0 +1,91 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/infineon,peb2466.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Infineon PEB2466 codec
+
+maintainers:
+ - Herve Codina <herve.codina@bootlin.com>
+
+description: |
+ The Infineon PEB2466 codec is a programmable DSP-based four channels codec
+ with filters capabilities.
+
+ The time-slots used by the codec must be set and so, the properties
+ 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and
+ 'dai-tdm-slot-rx-mask' must be present in the sound card node for sub-nodes
+ that involve the codec. The codec uses one 8bit time-slot per channel.
+ 'dai-tdm-tdm-slot-with' must be set to 8.
+
+ The PEB2466 codec also supports 28 gpios (signaling pins).
+
+allOf:
+ - $ref: /schemas/spi/spi-peripheral-props.yaml
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: infineon,peb2466
+
+ reg:
+ description:
+ SPI device address.
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Master clock
+
+ clock-names:
+ items:
+ - const: mclk
+
+ spi-max-frequency:
+ maximum: 8192000
+
+ reset-gpios:
+ description:
+ GPIO used to reset the device.
+ maxItems: 1
+
+ firmware-name:
+ $ref: /schemas/types.yaml#/definitions/string
+ description:
+ Filters coefficients file to load. If this property is omitted, internal
+ filters are disabled.
+
+ '#sound-dai-cells':
+ const: 0
+
+ '#gpio-cells':
+ const: 2
+
+ gpio-controller: true
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+ - gpio-controller
+ - '#gpio-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@0 {
+ compatible = "infineon,peb2466";
+ reg = <0>;
+ spi-max-frequency = <8192000>;
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+ #sound-dai-cells = <0>;
+ gpio-controller;
+ #gpio-cells = <2>;
+ };
+ };
diff --git a/Bindings/sound/ingenic,aic.yaml b/Bindings/sound/ingenic,aic.yaml
new file mode 100644
index 0000000..d15c000
--- /dev/null
+++ b/Bindings/sound/ingenic,aic.yaml
@@ -0,0 +1,90 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ingenic,aic.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Ingenic SoCs AC97 / I2S Controller (AIC)
+
+maintainers:
+ - Paul Cercueil <paul@crapouillou.net>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: '^audio-controller@'
+
+ compatible:
+ oneOf:
+ - enum:
+ - ingenic,jz4740-i2s
+ - ingenic,jz4760-i2s
+ - ingenic,jz4770-i2s
+ - ingenic,jz4780-i2s
+ - ingenic,x1000-i2s
+ - items:
+ - const: ingenic,jz4725b-i2s
+ - const: ingenic,jz4740-i2s
+
+ '#sound-dai-cells':
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: AIC clock
+ - description: I2S clock
+
+ clock-names:
+ items:
+ - const: aic
+ - const: i2s
+
+ dmas:
+ items:
+ - description: DMA controller phandle and request line for I2S RX
+ - description: DMA controller phandle and request line for I2S TX
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - '#sound-dai-cells'
+
+examples:
+ - |
+ #include <dt-bindings/clock/ingenic,jz4740-cgu.h>
+ aic: audio-controller@10020000 {
+ compatible = "ingenic,jz4740-i2s";
+ reg = <0x10020000 0x38>;
+
+ #sound-dai-cells = <0>;
+
+ interrupt-parent = <&intc>;
+ interrupts = <18>;
+
+ clocks = <&cgu JZ4740_CLK_AIC>,
+ <&cgu JZ4740_CLK_I2S>;
+ clock-names = "aic", "i2s";
+
+ dmas = <&dmac 25 0xffffffff>, <&dmac 24 0xffffffff>;
+ dma-names = "rx", "tx";
+ };
diff --git a/Bindings/sound/ingenic,codec.yaml b/Bindings/sound/ingenic,codec.yaml
new file mode 100644
index 0000000..b58b908
--- /dev/null
+++ b/Bindings/sound/ingenic,codec.yaml
@@ -0,0 +1,63 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ingenic,codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Ingenic JZ47xx internal codec
+
+maintainers:
+ - Paul Cercueil <paul@crapouillou.net>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: '^audio-codec@.*'
+
+ compatible:
+ oneOf:
+ - enum:
+ - ingenic,jz4770-codec
+ - ingenic,jz4760-codec
+ - ingenic,jz4725b-codec
+ - ingenic,jz4740-codec
+ - items:
+ - const: ingenic,jz4760b-codec
+ - const: ingenic,jz4760-codec
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: aic
+
+ '#sound-dai-cells':
+ const: 0
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - '#sound-dai-cells'
+
+examples:
+ - |
+ #include <dt-bindings/clock/ingenic,jz4740-cgu.h>
+ codec: audio-codec@10020080 {
+ compatible = "ingenic,jz4740-codec";
+ reg = <0x10020080 0x8>;
+ #sound-dai-cells = <0>;
+ clocks = <&cgu JZ4740_CLK_AIC>;
+ clock-names = "aic";
+ };
+
+...
diff --git a/Bindings/sound/inno-rk3036.txt b/Bindings/sound/inno-rk3036.txt
new file mode 100644
index 0000000..758de8e
--- /dev/null
+++ b/Bindings/sound/inno-rk3036.txt
@@ -0,0 +1,20 @@
+Inno audio codec for RK3036
+
+Inno audio codec is integrated inside RK3036 SoC.
+
+Required properties:
+- compatible : Should be "rockchip,rk3036-codec".
+- reg : The registers of codec.
+- clock-names : Should be "acodec_pclk".
+- clocks : The clock of codec.
+- rockchip,grf : The phandle of grf device node.
+
+Example:
+
+ acodec: acodec-ana@20030000 {
+ compatible = "rk3036-codec";
+ reg = <0x20030000 0x4000>;
+ rockchip,grf = <&grf>;
+ clock-names = "acodec_pclk";
+ clocks = <&cru ACLK_VCODEC>;
+ };
diff --git a/Bindings/sound/intel,keembay-i2s.yaml b/Bindings/sound/intel,keembay-i2s.yaml
new file mode 100644
index 0000000..76b6f2c
--- /dev/null
+++ b/Bindings/sound/intel,keembay-i2s.yaml
@@ -0,0 +1,90 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+# Copyright 2020 Intel Corporation
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Intel KeemBay I2S
+
+maintainers:
+ - Daniele Alessandrelli <daniele.alessandrelli@intel.com>
+ - Paul J. Murphy <paul.j.murphy@intel.com>
+
+description: |
+ Intel KeemBay I2S
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - intel,keembay-i2s
+ - intel,keembay-tdm
+ - intel,keembay-hdmi-i2s
+
+ "#sound-dai-cells":
+ const: 0
+
+ reg:
+ items:
+ - description: I2S registers
+ - description: I2S gen configuration
+
+ reg-names:
+ items:
+ - const: i2s-regs
+ - const: i2s_gen_cfg
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Bus Clock
+ - description: Module Clock
+
+ clock-names:
+ items:
+ - const: osc
+ - const: apb_clk
+
+ dmas:
+ items:
+ - description: DMA TX channel
+ - description: DMA RX channel
+
+ dma-names:
+ items:
+ - const: tx
+ - const: rx
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+ - reg
+ - clocks
+ - clock-names
+ - interrupts
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #define KEEM_BAY_PSS_AUX_I2S3
+ #define KEEM_BAY_PSS_I2S3
+ i2s3: i2s@20140000 {
+ compatible = "intel,keembay-i2s";
+ #sound-dai-cells = <0>;
+ reg = <0x20140000 0x200>, /* I2S registers */
+ <0x202a00a4 0x4>; /* I2S gen configuration */
+ reg-names = "i2s-regs", "i2s_gen_cfg";
+ interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>;
+ clock-names = "osc", "apb_clk";
+ clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>;
+ dmas = <&axi_dma0 29>, <&axi_dma0 33>;
+ dma-names = "tx", "rx";
+ };
diff --git a/Bindings/sound/irondevice,sma1303.yaml b/Bindings/sound/irondevice,sma1303.yaml
new file mode 100644
index 0000000..b36c35e
--- /dev/null
+++ b/Bindings/sound/irondevice,sma1303.yaml
@@ -0,0 +1,48 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/irondevice,sma1303.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Iron Device SMA1303 Audio Amplifier
+
+maintainers:
+ - Kiseok Jo <kiseok.jo@irondevice.com>
+
+description:
+ SMA1303 digital class-D audio amplifier
+ with an integrated boost converter.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - irondevice,sma1303
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 1
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ amplifier@1e {
+ compatible = "irondevice,sma1303";
+ reg = <0x1e>;
+ #sound-dai-cells = <1>;
+ };
+ };
diff --git a/Bindings/sound/linux,bt-sco.yaml b/Bindings/sound/linux,bt-sco.yaml
new file mode 100644
index 0000000..a67b79c
--- /dev/null
+++ b/Bindings/sound/linux,bt-sco.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Bluetooth SCO Audio Codec
+
+maintainers:
+ - Mark Brown <broonie@kernel.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ '#sound-dai-cells':
+ enum:
+ - 0
+
+ # For Wideband PCM
+ - 1
+
+ compatible:
+ enum:
+ - delta,dfbmcs320
+ - linux,bt-sco
+
+required:
+ - '#sound-dai-cells'
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ codec {
+ #sound-dai-cells = <0>;
+ compatible = "linux,bt-sco";
+ };
+
+...
diff --git a/Bindings/sound/linux,spdif-dit.yaml b/Bindings/sound/linux,spdif-dit.yaml
new file mode 100644
index 0000000..fe5f075
--- /dev/null
+++ b/Bindings/sound/linux,spdif-dit.yaml
@@ -0,0 +1,37 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/linux,spdif-dit.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Dummy SPDIF Transmitter
+
+maintainers:
+ - Mark Brown <broonie@kernel.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: linux,spdif-dit
+
+ "#sound-dai-cells":
+ const: 0
+
+ sound-name-prefix: true
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+
+additionalProperties: false
+
+examples:
+ - |
+ spdif-out {
+ #sound-dai-cells = <0>;
+ compatible = "linux,spdif-dit";
+ };
+
+...
diff --git a/Bindings/sound/loongson,ls-audio-card.yaml b/Bindings/sound/loongson,ls-audio-card.yaml
new file mode 100644
index 0000000..61e8bab
--- /dev/null
+++ b/Bindings/sound/loongson,ls-audio-card.yaml
@@ -0,0 +1,70 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Loongson 7axxx/2kxxx ASoC audio sound card driver
+
+maintainers:
+ - Yingkun Meng <mengyingkun@loongson.cn>
+
+description:
+ The binding describes the sound card present in loongson
+ 7axxx/2kxxx platform. The sound card is an ASoC component
+ which uses Loongson I2S controller to transfer the audio data.
+
+properties:
+ compatible:
+ const: loongson,ls-audio-card
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+ mclk-fs:
+ $ref: simple-card.yaml#/definitions/mclk-fs
+
+ cpu:
+ description: Holds subnode which indicates cpu dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ maxItems: 1
+ required:
+ - sound-dai
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ maxItems: 1
+ required:
+ - sound-dai
+
+required:
+ - compatible
+ - model
+ - mclk-fs
+ - cpu
+ - codec
+
+additionalProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "loongson,ls-audio-card";
+ model = "loongson-audio";
+ mclk-fs = <512>;
+
+ cpu {
+ sound-dai = <&i2s>;
+ };
+ codec {
+ sound-dai = <&es8323>;
+ };
+ };
diff --git a/Bindings/sound/marvell,mmp-sspa.yaml b/Bindings/sound/marvell,mmp-sspa.yaml
new file mode 100644
index 0000000..4193d17
--- /dev/null
+++ b/Bindings/sound/marvell,mmp-sspa.yaml
@@ -0,0 +1,105 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/marvell,mmp-sspa.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Marvel SSPA Digital Audio Interface
+
+maintainers:
+ - Lubomir Rintel <lkundrak@v3.sk>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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
+ additionalProperties: true
+
+ properties:
+ dai-format:
+ const: i2s
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - port
+
+unevaluatedProperties: 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/Bindings/sound/marvell,pxa2xx-ac97.txt b/Bindings/sound/marvell,pxa2xx-ac97.txt
new file mode 100644
index 0000000..2ea85d5
--- /dev/null
+++ b/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/Bindings/sound/max98373.txt b/Bindings/sound/max98373.txt
new file mode 100644
index 0000000..456cb1c
--- /dev/null
+++ b/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/Bindings/sound/max9860.txt b/Bindings/sound/max9860.txt
new file mode 100644
index 0000000..e0d4e95
--- /dev/null
+++ b/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/Bindings/sound/maxim,max9759.yaml b/Bindings/sound/maxim,max9759.yaml
new file mode 100644
index 0000000..a76ee6a
--- /dev/null
+++ b/Bindings/sound/maxim,max9759.yaml
@@ -0,0 +1,45 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max9759.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim MAX9759 Speaker Amplifier
+
+maintainers:
+ - Otabek Nazrullaev <otabeknazrullaev1998@gmail.com>
+
+properties:
+ compatible:
+ const: maxim,max9759
+
+ shutdown-gpios:
+ maxItems: 1
+ description: the gpio connected to the shutdown pin
+
+ mute-gpios:
+ maxItems: 1
+ description: the gpio connected to the mute pin
+
+ gain-gpios:
+ maxItems: 2
+ description: the 2 gpios connected to the g1 and g2 pins
+
+required:
+ - compatible
+ - shutdown-gpios
+ - mute-gpios
+ - gain-gpios
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ amplifier {
+ compatible = "maxim,max9759";
+ shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>;
+ mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>;
+ gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>,
+ <&gpio3 25 GPIO_ACTIVE_LOW>;
+ };
diff --git a/Bindings/sound/maxim,max98088.txt b/Bindings/sound/maxim,max98088.txt
new file mode 100644
index 0000000..da764d9
--- /dev/null
+++ b/Bindings/sound/maxim,max98088.txt
@@ -0,0 +1,23 @@
+MAX98088 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible: "maxim,max98088" or "maxim,max98089".
+- reg: The I2C address of the device.
+
+Optional properties:
+
+- clocks: the clock provider of MCLK, see ../clock/clock-bindings.txt section
+ "consumer" for more information.
+- clock-names: must be set to "mclk"
+
+Example:
+
+max98089: codec@10 {
+ compatible = "maxim,max98089";
+ reg = <0x10>;
+ clocks = <&clks IMX6QDL_CLK_CKO2>;
+ clock-names = "mclk";
+};
diff --git a/Bindings/sound/maxim,max98090.yaml b/Bindings/sound/maxim,max98090.yaml
new file mode 100644
index 0000000..65e4c51
--- /dev/null
+++ b/Bindings/sound/maxim,max98090.yaml
@@ -0,0 +1,84 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98090.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98090/MAX98091 audio codecs
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+description: |
+ Pins on the device (for linking into audio routes):
+ MIC1, MIC2, DMICL, DMICR, IN1, IN2, IN3, IN4, IN5, IN6, IN12, IN34, IN56,
+ HPL, HPR, SPKL, SPKR, RCVL, RCVR, MICBIAS
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - maxim,max98090
+ - maxim,max98091
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: master clock
+
+ clock-names:
+ items:
+ - const: mclk
+
+ interrupts:
+ maxItems: 1
+
+ maxim,dmic-freq:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ default: 2500000
+ description:
+ DMIC clock frequency
+
+ maxim,micbias:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [ 0, 1, 2, 3 ]
+ default: 3
+ description: |
+ Micbias voltage applied to the analog mic, valid voltages value are:
+ 0 - 2.2v
+ 1 - 2.55v
+ 2 - 2.4v
+ 3 - 2.8v
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - interrupts
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@10 {
+ compatible = "maxim,max98090";
+ reg = <0x10>;
+ interrupt-parent = <&gpx3>;
+ interrupts = <2 IRQ_TYPE_EDGE_FALLING>;
+ clocks = <&i2s0 0>;
+ clock-names = "mclk";
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/maxim,max98095.yaml b/Bindings/sound/maxim,max98095.yaml
new file mode 100644
index 0000000..77544a9
--- /dev/null
+++ b/Bindings/sound/maxim,max98095.yaml
@@ -0,0 +1,54 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98095.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98095 audio codec
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - maxim,max98095
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: master clock
+
+ clock-names:
+ items:
+ - const: mclk
+
+ '#sound-dai-cells':
+ const: 1
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@11 {
+ compatible = "maxim,max98095";
+ reg = <0x11>;
+ clocks = <&i2s0 0>;
+ clock-names = "mclk";
+ };
+ };
diff --git a/Bindings/sound/maxim,max98357a.yaml b/Bindings/sound/maxim,max98357a.yaml
new file mode 100644
index 0000000..83ba866
--- /dev/null
+++ b/Bindings/sound/maxim,max98357a.yaml
@@ -0,0 +1,52 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98357a.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98357A/MAX98360A amplifier
+
+maintainers:
+ - Tzung-Bi Shih <tzungbi@kernel.org>
+
+description:
+ Maxim Integrated MAX98357A/MAX98360A is a digital pulse-code modulation (PCM)
+ input Class D amplifier.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - maxim,max98357a
+ - maxim,max98360a
+
+ '#sound-dai-cells':
+ const: 0
+
+ sdmode-gpios:
+ maxItems: 1
+ description:
+ Chip's SD_MODE pin. If missing the chip is always on.
+
+ sdmode-delay:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Delay time for SD_MODE pin changes intended to make I2S clocks ready
+ before SD_MODE is unmuted in order to avoid the speaker pop noise.
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ amplifier {
+ compatible = "maxim,max98360a";
+ #sound-dai-cells = <0>;
+ sdmode-gpios = <&qcom_pinmux 25 GPIO_ACTIVE_HIGH>;
+ };
diff --git a/Bindings/sound/maxim,max98371.yaml b/Bindings/sound/maxim,max98371.yaml
new file mode 100644
index 0000000..14fba34
--- /dev/null
+++ b/Bindings/sound/maxim,max98371.yaml
@@ -0,0 +1,42 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98371.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim MAX98371 audio codec
+
+maintainers:
+ - anish kumar <yesanishhere@gmail.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: maxim,max98371
+
+ '#sound-dai-cells':
+ const: 0
+
+ reg:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@31 {
+ compatible = "maxim,max98371";
+ reg = <0x31>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/maxim,max98390.yaml b/Bindings/sound/maxim,max98390.yaml
new file mode 100644
index 0000000..deaa688
--- /dev/null
+++ b/Bindings/sound/maxim,max98390.yaml
@@ -0,0 +1,54 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1
+ maximum: 8388607
+
+ reset-gpios:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ max98390: amplifier@38 {
+ compatible = "maxim,max98390";
+ reg = <0x38>;
+ maxim,temperature_calib = <1024>;
+ maxim,r0_calib = <100232>;
+ reset-gpios = <&gpio 9 GPIO_ACTIVE_LOW>;
+ };
+ };
diff --git a/Bindings/sound/maxim,max98504.yaml b/Bindings/sound/maxim,max98504.yaml
new file mode 100644
index 0000000..23f19a9
--- /dev/null
+++ b/Bindings/sound/maxim,max98504.yaml
@@ -0,0 +1,86 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98504.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98504 class D mono speaker amplifier
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+description:
+ Maxim Integrated MAX98504 speaker amplifier supports I2C control interface
+ with an IRQ output signal, PCM and PDM digital audio interface (DAI) and a
+ differential analog input.
+
+properties:
+ compatible:
+ const: maxim,max98504
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ DIOVDD-supply: true
+ DVDD-supply: true
+ PVDD-supply: true
+
+ maxim,brownout-threshold:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ maximum: 21
+ default: 0
+ description:
+ PVDD brownout threshold, where values correspond to 2.6V, 2.65V...3.65V
+ voltage range. Property also enables the PVDD brownout protection.
+
+ maxim,brownout-attenuation:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ maximum: 6
+ default: 0
+ description:
+ Brownout attenuation to the speaker gain applied during the "attack hold"
+ and "timed hold" phase, the value must be from 0...6 (dB) range.
+
+ maxim,brownout-attack-hold-ms:
+ maximum: 255
+ default: 0
+ description:
+ Brownout attack hold phase time in ms, VBATBROWN_ATTK_HOLD, register 0x0018.
+
+ maxim,brownout-timed-hold-ms:
+ maximum: 255
+ default: 0
+ description:
+ Brownout timed hold phase time in ms, VBATBROWN_TIME_HOLD, register 0x0019.
+
+ maxim,brownout-release-rate-ms:
+ maximum: 255
+ default: 0
+ description:
+ Brownout release phase step time in ms, VBATBROWN_RELEASE, register 0x001A.
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ amplifier@31 {
+ compatible = "maxim,max98504";
+ reg = <0x31>;
+
+ DIOVDD-supply = <&ldo3_reg>;
+ DVDD-supply = <&ldo3_reg>;
+ };
+ };
diff --git a/Bindings/sound/maxim,max98520.yaml b/Bindings/sound/maxim,max98520.yaml
new file mode 100644
index 0000000..3f88c7d
--- /dev/null
+++ b/Bindings/sound/maxim,max98520.yaml
@@ -0,0 +1,35 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98520 Speaker Amplifier Driver
+
+maintainers:
+ - George Song <george.song@maximintegrated.com>
+
+properties:
+ compatible:
+ const: maxim,max98520
+
+ reg:
+ maxItems: 1
+ description: I2C address of the device.
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ max98520: amplifier@38 {
+ compatible = "maxim,max98520";
+ reg = <0x38>;
+ };
+ };
diff --git a/Bindings/sound/maxim,max9867.yaml b/Bindings/sound/maxim,max9867.yaml
new file mode 100644
index 0000000..0b9a84d
--- /dev/null
+++ b/Bindings/sound/maxim,max9867.yaml
@@ -0,0 +1,60 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max9867.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX9867 CODEC
+
+description: |
+ This device supports I2C only.
+ Pins on the device (for linking into audio routes):
+ * LOUT
+ * ROUT
+ * LINL
+ * LINR
+ * MICL
+ * MICR
+ * DMICL
+ * DMICR
+
+maintainers:
+ - Ladislav Michl <ladis@linux-mips.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - maxim,max9867
+
+ '#sound-dai-cells':
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - clocks
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@18 {
+ compatible = "maxim,max9867";
+ #sound-dai-cells = <0>;
+ reg = <0x18>;
+ clocks = <&codec_clk>;
+ };
+ };
+...
diff --git a/Bindings/sound/maxim,max98925.yaml b/Bindings/sound/maxim,max98925.yaml
new file mode 100644
index 0000000..32fd862
--- /dev/null
+++ b/Bindings/sound/maxim,max98925.yaml
@@ -0,0 +1,98 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier
+
+maintainers:
+ - Ryan Lee <ryans.lee@maximintegrated.com>
+
+properties:
+ compatible:
+ enum:
+ - maxim,max98925
+ - maxim,max98926
+ - maxim,max98927
+
+ reg:
+ maxItems: 1
+
+ reset-gpios:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ vmon-slot-no:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 30
+ default: 0
+ description:
+ Slot number used to send voltage information or in inteleave mode this
+ will be used as interleave slot.
+
+ imon-slot-no:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 0
+ maximum: 30
+ default: 0
+ description:
+ Slot number used to send current information.
+
+ maxim,interleave-mode:
+ type: boolean
+ description:
+ When using two MAX9892X in a system it is possible to create ADC data
+ that will overflow the frame size. When enabled, the Digital Audio
+ Interleave mode provides a means to output VMON and IMON data from two
+ devices on a single DOUT line when running smaller frames sizes such as
+ 32 BCLKS per LRCLK or 48 BCLKS per LRCLK.
+
+required:
+ - compatible
+ - reg
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - maxim,max98927
+ then:
+ properties:
+ vmon-slot-no:
+ minimum: 0
+ maximum: 15
+
+ imon-slot-no:
+ minimum: 0
+ maximum: 15
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ #include <dt-bindings/gpio/gpio.h>
+ audio-codec@3a {
+ compatible = "maxim,max98927";
+ reg = <0x3a>;
+ #sound-dai-cells = <0>;
+
+ pinctrl-0 = <&speaker_default>;
+ pinctrl-names = "default";
+
+ reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>;
+
+ vmon-slot-no = <1>;
+ imon-slot-no = <0>;
+ };
+ };
diff --git a/Bindings/sound/mediatek,mt7986-afe.yaml b/Bindings/sound/mediatek,mt7986-afe.yaml
new file mode 100644
index 0000000..398efdf
--- /dev/null
+++ b/Bindings/sound/mediatek,mt7986-afe.yaml
@@ -0,0 +1,160 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt7986-afe.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek AFE PCM controller for MT7986
+
+maintainers:
+ - Maso Huang <maso.huang@mediatek.com>
+
+properties:
+ compatible:
+ oneOf:
+ - const: mediatek,mt7986-afe
+ - items:
+ - enum:
+ - mediatek,mt7981-afe
+ - mediatek,mt7988-afe
+ - const: mediatek,mt7986-afe
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ minItems: 5
+ items:
+ - description: audio bus clock
+ - description: audio 26M clock
+ - description: audio intbus clock
+ - description: audio hopping clock
+ - description: audio pll clock
+ - description: mux for pcm_mck
+ - description: audio i2s/pcm mck
+
+ clock-names:
+ minItems: 5
+ items:
+ - const: bus_ck
+ - const: 26m_ck
+ - const: l_ck
+ - const: aud_ck
+ - const: eg2_ck
+ - const: sel
+ - const: i2s_m
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+
+allOf:
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: mediatek,mt7986-afe
+ then:
+ properties:
+ clocks:
+ items:
+ - description: audio bus clock
+ - description: audio 26M clock
+ - description: audio intbus clock
+ - description: audio hopping clock
+ - description: audio pll clock
+ clock-names:
+ items:
+ - const: bus_ck
+ - const: 26m_ck
+ - const: l_ck
+ - const: aud_ck
+ - const: eg2_ck
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: mediatek,mt7981-afe
+ then:
+ properties:
+ clocks:
+ items:
+ - description: audio bus clock
+ - description: audio 26M clock
+ - description: audio intbus clock
+ - description: audio hopping clock
+ - description: audio pll clock
+ - description: mux for pcm_mck
+ clock-names:
+ items:
+ - const: bus_ck
+ - const: 26m_ck
+ - const: l_ck
+ - const: aud_ck
+ - const: eg2_ck
+ - const: sel
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: mediatek,mt7988-afe
+ then:
+ properties:
+ clocks:
+ items:
+ - description: audio bus clock
+ - description: audio 26M clock
+ - description: audio intbus clock
+ - description: audio hopping clock
+ - description: audio pll clock
+ - description: mux for pcm_mck
+ - description: audio i2s/pcm mck
+ clock-names:
+ items:
+ - const: bus_ck
+ - const: 26m_ck
+ - const: l_ck
+ - const: aud_ck
+ - const: eg2_ck
+ - const: sel
+ - const: i2s_m
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/clock/mt7986-clk.h>
+
+ afe@11210000 {
+ compatible = "mediatek,mt7986-afe";
+ reg = <0x11210000 0x9000>;
+ interrupts = <GIC_SPI 106 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&infracfg_ao CLK_INFRA_AUD_BUS_CK>,
+ <&infracfg_ao CLK_INFRA_AUD_26M_CK>,
+ <&infracfg_ao CLK_INFRA_AUD_L_CK>,
+ <&infracfg_ao CLK_INFRA_AUD_AUD_CK>,
+ <&infracfg_ao CLK_INFRA_AUD_EG2_CK>;
+ clock-names = "bus_ck",
+ "26m_ck",
+ "l_ck",
+ "aud_ck",
+ "eg2_ck";
+ assigned-clocks = <&topckgen CLK_TOP_A1SYS_SEL>,
+ <&topckgen CLK_TOP_AUD_L_SEL>,
+ <&topckgen CLK_TOP_A_TUNER_SEL>;
+ assigned-clock-parents = <&topckgen CLK_TOP_APLL2_D4>,
+ <&apmixedsys CLK_APMIXED_APLL2>,
+ <&topckgen CLK_TOP_APLL2_D4>;
+ };
+
+...
diff --git a/Bindings/sound/mediatek,mt7986-wm8960.yaml b/Bindings/sound/mediatek,mt7986-wm8960.yaml
new file mode 100644
index 0000000..09247ce
--- /dev/null
+++ b/Bindings/sound/mediatek,mt7986-wm8960.yaml
@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt7986-wm8960.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek MT7986 sound card with WM8960 codec
+
+maintainers:
+ - Maso Huang <maso.huang@mediatek.com>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ const: mediatek,mt7986-wm8960-sound
+
+ platform:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ description: The phandle of MT7986 platform.
+ maxItems: 1
+ required:
+ - sound-dai
+
+ codec:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ description: The phandle of wm8960 codec.
+ maxItems: 1
+ required:
+ - sound-dai
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - audio-routing
+ - platform
+ - codec
+
+examples:
+ - |
+ sound {
+ compatible = "mediatek,mt7986-wm8960-sound";
+ model = "mt7986-wm8960";
+ audio-routing =
+ "Headphone", "HP_L",
+ "Headphone", "HP_R",
+ "LINPUT1", "AMIC",
+ "RINPUT1", "AMIC";
+
+ platform {
+ sound-dai = <&afe>;
+ };
+
+ codec {
+ sound-dai = <&wm8960>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/mediatek,mt8188-afe.yaml b/Bindings/sound/mediatek,mt8188-afe.yaml
new file mode 100644
index 0000000..77af276
--- /dev/null
+++ b/Bindings/sound/mediatek,mt8188-afe.yaml
@@ -0,0 +1,241 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt8188-afe.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek AFE PCM controller for mt8188
+
+maintainers:
+ - Trevor Wu <trevor.wu@mediatek.com>
+
+properties:
+ compatible:
+ const: mediatek,mt8188-afe
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ reset-names:
+ const: audiosys
+
+ memory-region:
+ maxItems: 1
+ description: |
+ Shared memory region for AFE memif. A "shared-dma-pool".
+ See dtschema reserved-memory/shared-dma-pool.yaml for details.
+
+ mediatek,topckgen:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the mediatek topckgen controller
+
+ mediatek,infracfg:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the mediatek infracfg controller
+
+ power-domains:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: 26M clock
+ - description: audio pll1 clock
+ - description: audio pll2 clock
+ - description: clock divider for i2si1_mck
+ - description: clock divider for i2si2_mck
+ - description: clock divider for i2so1_mck
+ - description: clock divider for i2so2_mck
+ - description: clock divider for dptx_mck
+ - description: a1sys hoping clock
+ - description: audio intbus clock
+ - description: audio hires clock
+ - description: audio local bus clock
+ - description: mux for dptx_mck
+ - description: mux for i2so1_mck
+ - description: mux for i2so2_mck
+ - description: mux for i2si1_mck
+ - description: mux for i2si2_mck
+ - description: audio 26m clock
+ - description: audio pll1 divide 4
+ - description: audio pll2 divide 4
+ - description: clock divider for iec
+ - description: mux for a2sys clock
+ - description: mux for aud_iec
+
+ clock-names:
+ items:
+ - const: clk26m
+ - const: apll1
+ - const: apll2
+ - const: apll12_div0
+ - const: apll12_div1
+ - const: apll12_div2
+ - const: apll12_div3
+ - const: apll12_div9
+ - const: top_a1sys_hp
+ - const: top_aud_intbus
+ - const: top_audio_h
+ - const: top_audio_local_bus
+ - const: top_dptx
+ - const: top_i2so1
+ - const: top_i2so2
+ - const: top_i2si1
+ - const: top_i2si2
+ - const: adsp_audio_26m
+ - const: apll1_d4
+ - const: apll2_d4
+ - const: apll12_div4
+ - const: top_a2sys
+ - const: top_aud_iec
+
+ mediatek,etdm-in1-cowork-source:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ etdm modules can share the same external clock pin. Specify
+ which etdm clock source is required by this etdm in module.
+ enum:
+ - 1 # etdm2_in
+ - 2 # etdm1_out
+ - 3 # etdm2_out
+
+ mediatek,etdm-in2-cowork-source:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ etdm modules can share the same external clock pin. Specify
+ which etdm clock source is required by this etdm in module.
+ enum:
+ - 0 # etdm1_in
+ - 2 # etdm1_out
+ - 3 # etdm2_out
+
+ mediatek,etdm-out1-cowork-source:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ etdm modules can share the same external clock pin. Specify
+ which etdm clock source is required by this etdm out module.
+ enum:
+ - 0 # etdm1_in
+ - 1 # etdm2_in
+ - 3 # etdm2_out
+
+ mediatek,etdm-out2-cowork-source:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ etdm modules can share the same external clock pin. Specify
+ which etdm clock source is required by this etdm out module.
+ enum:
+ - 0 # etdm1_in
+ - 1 # etdm2_in
+ - 2 # etdm1_out
+
+patternProperties:
+ "^mediatek,etdm-in[1-2]-chn-disabled$":
+ $ref: /schemas/types.yaml#/definitions/uint8-array
+ minItems: 1
+ maxItems: 16
+ description:
+ This is a list of channel IDs which should be disabled.
+ By default, all data received from ETDM pins will be outputted to
+ memory. etdm in supports disable_out in direct mode(w/o interconn),
+ so user can disable the specified channels by the property.
+ uniqueItems: true
+ items:
+ minimum: 0
+ maximum: 15
+
+ "^mediatek,etdm-in[1-2]-multi-pin-mode$":
+ type: boolean
+ description: if present, the etdm data mode is I2S.
+
+ "^mediatek,etdm-out[1-3]-multi-pin-mode$":
+ type: boolean
+ description: if present, the etdm data mode is I2S.
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - resets
+ - reset-names
+ - mediatek,topckgen
+ - mediatek,infracfg
+ - power-domains
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ afe@10b10000 {
+ compatible = "mediatek,mt8188-afe";
+ reg = <0x10b10000 0x10000>;
+ interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>;
+ resets = <&watchdog 14>;
+ reset-names = "audiosys";
+ memory-region = <&snd_dma_mem_reserved>;
+ mediatek,topckgen = <&topckgen>;
+ mediatek,infracfg = <&infracfg_ao>;
+ power-domains = <&spm 13>; //MT8188_POWER_DOMAIN_AUDIO
+ mediatek,etdm-in2-cowork-source = <2>;
+ mediatek,etdm-out2-cowork-source = <0>;
+ mediatek,etdm-in1-multi-pin-mode;
+ mediatek,etdm-in1-chn-disabled = /bits/ 8 <0x0 0x2>;
+ clocks = <&clk26m>,
+ <&apmixedsys 9>, //CLK_APMIXED_APLL1
+ <&apmixedsys 10>, //CLK_APMIXED_APLL2
+ <&topckgen 186>, //CLK_TOP_APLL12_CK_DIV0
+ <&topckgen 187>, //CLK_TOP_APLL12_CK_DIV1
+ <&topckgen 188>, //CLK_TOP_APLL12_CK_DIV2
+ <&topckgen 189>, //CLK_TOP_APLL12_CK_DIV3
+ <&topckgen 191>, //CLK_TOP_APLL12_CK_DIV9
+ <&topckgen 83>, //CLK_TOP_A1SYS_HP
+ <&topckgen 31>, //CLK_TOP_AUD_INTBUS
+ <&topckgen 32>, //CLK_TOP_AUDIO_H
+ <&topckgen 69>, //CLK_TOP_AUDIO_LOCAL_BUS
+ <&topckgen 81>, //CLK_TOP_DPTX
+ <&topckgen 77>, //CLK_TOP_I2SO1
+ <&topckgen 78>, //CLK_TOP_I2SO2
+ <&topckgen 79>, //CLK_TOP_I2SI1
+ <&topckgen 80>, //CLK_TOP_I2SI2
+ <&adsp_audio26m 0>, //CLK_AUDIODSP_AUDIO26M
+ <&topckgen 132>, //CLK_TOP_APLL1_D4
+ <&topckgen 133>, //CLK_TOP_APLL2_D4
+ <&topckgen 183>, //CLK_TOP_APLL12_CK_DIV4
+ <&topckgen 84>, //CLK_TOP_A2SYS
+ <&topckgen 82>; //CLK_TOP_AUD_IEC>;
+ clock-names = "clk26m",
+ "apll1",
+ "apll2",
+ "apll12_div0",
+ "apll12_div1",
+ "apll12_div2",
+ "apll12_div3",
+ "apll12_div9",
+ "top_a1sys_hp",
+ "top_aud_intbus",
+ "top_audio_h",
+ "top_audio_local_bus",
+ "top_dptx",
+ "top_i2so1",
+ "top_i2so2",
+ "top_i2si1",
+ "top_i2si2",
+ "adsp_audio_26m",
+ "apll1_d4",
+ "apll2_d4",
+ "apll12_div4",
+ "top_a2sys",
+ "top_aud_iec";
+ };
+
+...
diff --git a/Bindings/sound/mediatek,mt8188-mt6359.yaml b/Bindings/sound/mediatek,mt8188-mt6359.yaml
new file mode 100644
index 0000000..4c8c950
--- /dev/null
+++ b/Bindings/sound/mediatek,mt8188-mt6359.yaml
@@ -0,0 +1,114 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt8188-mt6359.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek MT8188 ASoC sound card
+
+maintainers:
+ - Trevor Wu <trevor.wu@mediatek.com>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - mediatek,mt8188-mt6359-evb
+ - mediatek,mt8188-nau8825
+ - mediatek,mt8188-rt5682s
+
+ audio-routing:
+ description:
+ Valid names could be the input or output widgets of audio components,
+ power supplies, MicBias of codec and the software switch.
+
+ mediatek,platform:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of MT8188 ASoC platform.
+
+patternProperties:
+ "^dai-link-[0-9]+$":
+ type: object
+ description:
+ Container for dai-link level properties and CODEC sub-nodes.
+
+ properties:
+ link-name:
+ description:
+ This property corresponds to the name of the BE dai-link to which
+ we are going to update parameters in this node.
+ items:
+ enum:
+ - DPTX_BE
+ - ETDM1_IN_BE
+ - ETDM2_IN_BE
+ - ETDM1_OUT_BE
+ - ETDM2_OUT_BE
+ - ETDM3_OUT_BE
+ - PCM1_BE
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+ required:
+ - sound-dai
+
+ dai-format:
+ description: audio format.
+ items:
+ enum:
+ - i2s
+ - right_j
+ - left_j
+ - dsp_a
+ - dsp_b
+
+ mediatek,clk-provider:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: Indicates dai-link clock master.
+ items:
+ enum:
+ - cpu
+ - codec
+
+ additionalProperties: false
+
+ required:
+ - link-name
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - mediatek,platform
+
+examples:
+ - |
+ sound {
+ compatible = "mediatek,mt8188-mt6359-evb";
+ model = "MT6359-EVB";
+ mediatek,platform = <&afe>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&aud_pins_default>;
+ audio-routing =
+ "Headphone", "Headphone L",
+ "Headphone", "Headphone R",
+ "AIN1", "Headset Mic";
+ dai-link-0 {
+ link-name = "ETDM3_OUT_BE";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&hdmi0>;
+ };
+ };
+ };
+
+...
diff --git a/Bindings/sound/microchip,sama7g5-i2smcc.yaml b/Bindings/sound/microchip,sama7g5-i2smcc.yaml
new file mode 100644
index 0000000..651f61c
--- /dev/null
+++ b/Bindings/sound/microchip,sama7g5-i2smcc.yaml
@@ -0,0 +1,110 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-i2smcc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Microchip I2S Multi-Channel Controller
+
+maintainers:
+ - Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
+
+description:
+ The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and
+ supports a Time Division Multiplexed (TDM) interface with external
+ multi-channel audio codecs. It consists of a receiver, a transmitter and a
+ common clock generator that can be enabled separately to provide Adapter,
+ Client or Controller modes with receiver and/or transmitter active.
+ On later I2SMCC versions (starting with Microchip's SAMA7G5) I2S
+ multi-channel is supported by using multiple data pins, output and
+ input, without TDM.
+
+properties:
+ "#sound-dai-cells":
+ const: 0
+
+ compatible:
+ 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
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ const: microchip,sam9x60-i2smcc
+ then:
+ properties:
+ microchip,tdm-data-pair: false
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/dma/at91.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+ i2s@f001c000 {
+ #sound-dai-cells = <0>;
+ compatible = "microchip,sam9x60-i2smcc";
+ reg = <0xf001c000 0x100>;
+ interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>;
+ dmas = <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
+ AT91_XDMAC_DT_PERID(36))>,
+ <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
+ AT91_XDMAC_DT_PERID(37))>;
+ dma-names = "tx", "rx";
+ clocks = <&i2s_clk>, <&i2s_gclk>;
+ clock-names = "pclk", "gclk";
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_i2s_default>;
+ };
diff --git a/Bindings/sound/microchip,sama7g5-pdmc.yaml b/Bindings/sound/microchip,sama7g5-pdmc.yaml
new file mode 100644
index 0000000..9aa65c9
--- /dev/null
+++ b/Bindings/sound/microchip,sama7g5-pdmc.yaml
@@ -0,0 +1,105 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-pdmc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Microchip Pulse Density Microphone Controller
+
+maintainers:
+ - Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
+
+description:
+ The Microchip Pulse Density Microphone Controller (PDMC) interfaces up to 4
+ digital microphones having Pulse Density Modulated (PDM) outputs.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: microchip,sama7g5-pdmc
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Peripheral Bus Clock
+ - description: Generic Clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: gclk
+
+ dmas:
+ description: RX DMA Channel
+ maxItems: 1
+
+ dma-names:
+ const: rx
+
+ microchip,mic-pos:
+ description: |
+ Position of PDM microphones on the DS line and the sampling edge (rising
+ or falling) of the CLK line. A microphone is represented as a pair of DS
+ line and the sampling edge. The first microphone is mapped to channel 0,
+ the second to channel 1, etc.
+ $ref: /schemas/types.yaml#/definitions/uint32-matrix
+ items:
+ items:
+ - description: value for DS line
+ enum: [0, 1]
+ - description: value for sampling edge
+ enum: [0, 1]
+ minItems: 1
+ maxItems: 4
+ uniqueItems: true
+
+ microchip,startup-delay-us:
+ description: |
+ Specifies the delay in microseconds that needs to be applied after
+ enabling the PDMC microphones to avoid unwanted noise due to microphones
+ not being ready.
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - microchip,mic-pos
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/at91.h>
+ #include <dt-bindings/dma/at91.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/sound/microchip,pdmc.h>
+
+ pdmc: sound@e1608000 {
+ compatible = "microchip,sama7g5-pdmc";
+ reg = <0xe1608000 0x4000>;
+ #sound-dai-cells = <0>;
+ interrupts = <GIC_SPI 68 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&dma0 AT91_XDMAC_DT_PERID(37)>;
+ dma-names = "rx";
+ clocks = <&pmc PMC_TYPE_PERIPHERAL 68>, <&pmc PMC_TYPE_GCK 68>;
+ clock-names = "pclk", "gclk";
+ microchip,mic-pos = <MCHP_PDMC_DS0 MCHP_PDMC_CLK_POSITIVE>,
+ <MCHP_PDMC_DS0 MCHP_PDMC_CLK_NEGATIVE>,
+ <MCHP_PDMC_DS1 MCHP_PDMC_CLK_POSITIVE>,
+ <MCHP_PDMC_DS1 MCHP_PDMC_CLK_NEGATIVE>;
+ };
diff --git a/Bindings/sound/microchip,sama7g5-spdifrx.yaml b/Bindings/sound/microchip,sama7g5-spdifrx.yaml
new file mode 100644
index 0000000..2f43c68
--- /dev/null
+++ b/Bindings/sound/microchip,sama7g5-spdifrx.yaml
@@ -0,0 +1,73 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdifrx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Microchip S/PDIF Rx Controller
+
+maintainers:
+ - Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
+
+description:
+ The Microchip Sony/Philips Digital Interface Receiver is a serial port
+ compliant with the IEC-60958 standard.
+
+properties:
+ "#sound-dai-cells":
+ const: 0
+
+ compatible:
+ const: microchip,sama7g5-spdifrx
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Peripheral Bus Clock
+ - description: Generic Clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: gclk
+
+ dmas:
+ description: RX DMA Channel
+ maxItems: 1
+
+ dma-names:
+ const: rx
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/at91.h>
+ #include <dt-bindings/dma/at91.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+ spdifrx: spdifrx@e1614000 {
+ #sound-dai-cells = <0>;
+ compatible = "microchip,sama7g5-spdifrx";
+ reg = <0xe1614000 0x4000>;
+ interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&dma0 AT91_XDMAC_DT_PERID(49)>;
+ dma-names = "rx";
+ clocks = <&pmc PMC_TYPE_PERIPHERAL 84>, <&pmc PMC_TYPE_GCK 84>;
+ clock-names = "pclk", "gclk";
+ };
diff --git a/Bindings/sound/microchip,sama7g5-spdiftx.yaml b/Bindings/sound/microchip,sama7g5-spdiftx.yaml
new file mode 100644
index 0000000..4702c52
--- /dev/null
+++ b/Bindings/sound/microchip,sama7g5-spdiftx.yaml
@@ -0,0 +1,78 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdiftx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Microchip S/PDIF Tx Controller
+
+maintainers:
+ - Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
+
+description:
+ The Microchip Sony/Philips Digital Interface Transmitter is a serial port
+ compliant with the IEC-60958 standard.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ "#sound-dai-cells":
+ const: 0
+
+ compatible:
+ const: microchip,sama7g5-spdiftx
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Peripheral Bus Clock
+ - description: Generic Clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: gclk
+
+ dmas:
+ description: TX DMA Channel
+ maxItems: 1
+
+ dma-names:
+ const: tx
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/at91.h>
+ #include <dt-bindings/dma/at91.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+ spdiftx@e1618000 {
+ #sound-dai-cells = <0>;
+ compatible = "microchip,sama7g5-spdiftx";
+ reg = <0xe1618000 0x4000>;
+ interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>;
+ dma-names = "tx";
+ clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>;
+ clock-names = "pclk", "gclk";
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_spdiftx_default>;
+ };
diff --git a/Bindings/sound/mikroe,mikroe-proto.txt b/Bindings/sound/mikroe,mikroe-proto.txt
new file mode 100644
index 0000000..912f8fa
--- /dev/null
+++ b/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/Bindings/sound/mrvl,pxa-ssp.txt b/Bindings/sound/mrvl,pxa-ssp.txt
new file mode 100644
index 0000000..feef39b
--- /dev/null
+++ b/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/Bindings/sound/mt2701-afe-pcm.txt b/Bindings/sound/mt2701-afe-pcm.txt
new file mode 100644
index 0000000..f548e6a
--- /dev/null
+++ b/Bindings/sound/mt2701-afe-pcm.txt
@@ -0,0 +1,146 @@
+Mediatek AFE PCM controller for mt2701
+
+Required properties:
+- compatible: should be one of the following.
+ - "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/Bindings/sound/mt2701-cs42448.txt b/Bindings/sound/mt2701-cs42448.txt
new file mode 100644
index 0000000..0557444
--- /dev/null
+++ b/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/Bindings/sound/mt2701-wm8960.txt b/Bindings/sound/mt2701-wm8960.txt
new file mode 100644
index 0000000..809b609
--- /dev/null
+++ b/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/Bindings/sound/mt6351.txt b/Bindings/sound/mt6351.txt
new file mode 100644
index 0000000..7fb2cb9
--- /dev/null
+++ b/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/Bindings/sound/mt6358.txt b/Bindings/sound/mt6358.txt
new file mode 100644
index 0000000..fbe9e55
--- /dev/null
+++ b/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/Bindings/sound/mt6359.yaml b/Bindings/sound/mt6359.yaml
new file mode 100644
index 0000000..23d411f
--- /dev/null
+++ b/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/Bindings/sound/mt6797-afe-pcm.txt b/Bindings/sound/mt6797-afe-pcm.txt
new file mode 100644
index 0000000..0ae29de
--- /dev/null
+++ b/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/Bindings/sound/mt6797-mt6351.txt b/Bindings/sound/mt6797-mt6351.txt
new file mode 100644
index 0000000..1d95a88
--- /dev/null
+++ b/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/Bindings/sound/mt8173-max98090.txt b/Bindings/sound/mt8173-max98090.txt
new file mode 100644
index 0000000..519e97c
--- /dev/null
+++ b/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/Bindings/sound/mt8173-rt5650-rt5514.txt b/Bindings/sound/mt8173-rt5650-rt5514.txt
new file mode 100644
index 0000000..e8b3c80
--- /dev/null
+++ b/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/Bindings/sound/mt8173-rt5650-rt5676.txt b/Bindings/sound/mt8173-rt5650-rt5676.txt
new file mode 100644
index 0000000..ac28cdb
--- /dev/null
+++ b/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/Bindings/sound/mt8173-rt5650.txt b/Bindings/sound/mt8173-rt5650.txt
new file mode 100644
index 0000000..29dce2a
--- /dev/null
+++ b/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/Bindings/sound/mt8183-afe-pcm.txt b/Bindings/sound/mt8183-afe-pcm.txt
new file mode 100644
index 0000000..1f1cba4
--- /dev/null
+++ b/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/Bindings/sound/mt8183-da7219-max98357.txt b/Bindings/sound/mt8183-da7219-max98357.txt
new file mode 100644
index 0000000..f276dfc
--- /dev/null
+++ b/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/Bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Bindings/sound/mt8183-mt6358-ts3a227-max98357.txt
new file mode 100644
index 0000000..ecd46ed
--- /dev/null
+++ b/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/Bindings/sound/mt8186-afe-pcm.yaml b/Bindings/sound/mt8186-afe-pcm.yaml
new file mode 100644
index 0000000..7fe85b0
--- /dev/null
+++ b/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/Bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Bindings/sound/mt8186-mt6366-da7219-max98357.yaml
new file mode 100644
index 0000000..9853c11
--- /dev/null
+++ b/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:
+ $ref: /schemas/types.yaml#/definitions/string-array
+ description:
+ A list of the desired dai-links in the sound card. Each entry is a
+ name defined in the machine driver.
+
+additionalProperties: false
+
+required:
+ - compatible
+ - mediatek,platform
+ - headset-codec
+ - playback-codecs
+
+examples:
+ - |
+
+ sound: mt8186-sound {
+ compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound";
+ mediatek,platform = <&afe>;
+ pinctrl-names = "aud_clk_mosi_off",
+ "aud_clk_mosi_on";
+ pinctrl-0 = <&aud_clk_mosi_off>;
+ pinctrl-1 = <&aud_clk_mosi_on>;
+
+ headset-codec {
+ sound-dai = <&da7219>;
+ };
+
+ playback-codecs {
+ sound-dai = <&anx_bridge_dp>,
+ <&max98357a>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
new file mode 100644
index 0000000..bdf7b09
--- /dev/null
+++ b/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
@@ -0,0 +1,99 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mt8186-mt6366-rt1019-rt5682s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Mediatek MT8186 with MT6366, RT1019 and RT5682S ASoC sound card driver
+
+maintainers:
+ - Jiaxin Yu <jiaxin.yu@mediatek.com>
+
+description:
+ This binding describes the MT8186 sound card.
+
+properties:
+ compatible:
+ enum:
+ - mediatek,mt8186-mt6366-rt1019-rt5682s-sound
+ - mediatek,mt8186-mt6366-rt5682s-max98360-sound
+ - mediatek,mt8186-mt6366-rt5650-sound
+
+ mediatek,platform:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of MT8186 ASoC platform.
+
+ dmic-gpios:
+ maxItems: 1
+ description:
+ dmic-gpios optional prop for switching between two DMICs.
+ Ex, the GPIO can control a MUX HW component to select
+ dmic clk and data form a Front or Rear dmic.
+
+ headset-codec:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ maxItems: 1
+ required:
+ - sound-dai
+
+ playback-codecs:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ items:
+ - description: phandle of dp codec
+ - description: phandle of l channel speaker codec
+ - description: phandle of r channel speaker codec
+ minItems: 2
+ required:
+ - sound-dai
+
+ mediatek,adsp:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of MT8186 ADSP platform.
+
+ mediatek,dai-link:
+ $ref: /schemas/types.yaml#/definitions/string-array
+ description:
+ A list of the desired dai-links in the sound card. Each entry is a
+ name defined in the machine driver.
+
+additionalProperties: false
+
+required:
+ - compatible
+ - mediatek,platform
+ - headset-codec
+ - playback-codecs
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ sound: mt8186-sound {
+ compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound";
+ mediatek,platform = <&afe>;
+ pinctrl-names = "aud_clk_mosi_off",
+ "aud_clk_mosi_on",
+ "aud_gpio_dmic_sec";
+ pinctrl-0 = <&aud_clk_mosi_off>;
+ pinctrl-1 = <&aud_clk_mosi_on>;
+ pinctrl-2 = <&aud_gpio_dmic_sec>;
+
+ dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>;
+
+ headset-codec {
+ sound-dai = <&rt5682s>;
+ };
+
+ playback-codecs {
+ sound-dai = <&it6505dptx>,
+ <&rt1019p>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/mt8192-afe-pcm.yaml b/Bindings/sound/mt8192-afe-pcm.yaml
new file mode 100644
index 0000000..064ef17
--- /dev/null
+++ b/Bindings/sound/mt8192-afe-pcm.yaml
@@ -0,0 +1,100 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the mediatek topckgen controller
+
+ power-domains:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: AFE clock
+ - description: ADDA DAC clock
+ - description: ADDA DAC pre-distortion clock
+ - description: audio infra sys clock
+ - description: audio infra 26M clock
+
+ clock-names:
+ items:
+ - const: aud_afe_clk
+ - const: aud_dac_clk
+ - const: aud_dac_predis_clk
+ - const: aud_infra_clk
+ - const: aud_infra_26m_clk
+
+required:
+ - compatible
+ - interrupts
+ - resets
+ - reset-names
+ - mediatek,apmixedsys
+ - mediatek,infracfg
+ - mediatek,topckgen
+ - power-domains
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/mt8192-clk.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/power/mt8192-power.h>
+ #include <dt-bindings/reset/mt8192-resets.h>
+
+ afe: mt8192-afe-pcm {
+ compatible = "mediatek,mt8192-audio";
+ interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>;
+ resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>;
+ reset-names = "audiosys";
+ mediatek,apmixedsys = <&apmixedsys>;
+ mediatek,infracfg = <&infracfg>;
+ mediatek,topckgen = <&topckgen>;
+ power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>;
+ clocks = <&audsys CLK_AUD_AFE>,
+ <&audsys CLK_AUD_DAC>,
+ <&audsys CLK_AUD_DAC_PREDIS>,
+ <&infracfg CLK_INFRA_AUDIO>,
+ <&infracfg CLK_INFRA_AUDIO_26M_B>;
+ clock-names = "aud_afe_clk",
+ "aud_dac_clk",
+ "aud_dac_predis_clk",
+ "aud_infra_clk",
+ "aud_infra_26m_clk";
+ };
+
+...
diff --git a/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
new file mode 100644
index 0000000..7e50f5d
--- /dev/null
+++ b/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
@@ -0,0 +1,84 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mt8192-mt6359-rt1015-rt5682.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Mediatek MT8192 with MT6359, RT1015 and RT5682 ASoC sound card driver
+
+maintainers:
+ - Jiaxin Yu <jiaxin.yu@mediatek.com>
+ - Shane Chien <shane.chien@mediatek.com>
+
+description:
+ This binding describes the MT8192 sound card.
+
+properties:
+ compatible:
+ enum:
+ - mediatek,mt8192_mt6359_rt1015_rt5682
+ - mediatek,mt8192_mt6359_rt1015p_rt5682
+ - mediatek,mt8192_mt6359_rt1015p_rt5682s
+
+ mediatek,platform:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of MT8192 ASoC platform.
+
+ mediatek,hdmi-codec:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of HDMI codec.
+
+ headset-codec:
+ type: object
+ additionalProperties: false
+
+ properties:
+ sound-dai:
+ maxItems: 1
+ required:
+ - sound-dai
+
+ speaker-codecs:
+ type: object
+ additionalProperties: false
+
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+ items:
+ maxItems: 1
+ required:
+ - sound-dai
+
+additionalProperties: false
+
+required:
+ - compatible
+ - mediatek,platform
+ - headset-codec
+ - speaker-codecs
+
+examples:
+ - |
+
+ sound: mt8192-sound {
+ compatible = "mediatek,mt8192_mt6359_rt1015_rt5682";
+ mediatek,platform = <&afe>;
+ mediatek,hdmi-codec = <&anx_bridge_dp>;
+ pinctrl-names = "aud_clk_mosi_off",
+ "aud_clk_mosi_on";
+ pinctrl-0 = <&aud_clk_mosi_off>;
+ pinctrl-1 = <&aud_clk_mosi_on>;
+
+ headset-codec {
+ sound-dai = <&rt5682>;
+ };
+
+ speaker-codecs {
+ sound-dai = <&rt1015_l>,
+ <&rt1015_r>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/mt8195-afe-pcm.yaml b/Bindings/sound/mt8195-afe-pcm.yaml
new file mode 100644
index 0000000..5c8dba2
--- /dev/null
+++ b/Bindings/sound/mt8195-afe-pcm.yaml
@@ -0,0 +1,200 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mt8195-afe-pcm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Mediatek AFE PCM controller for mt8195
+
+maintainers:
+ - Trevor Wu <trevor.wu@mediatek.com>
+
+properties:
+ compatible:
+ const: mediatek,mt8195-audio
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ reset-names:
+ const: audiosys
+
+ memory-region:
+ maxItems: 1
+ description: |
+ Shared memory region for AFE memif. A "shared-dma-pool".
+ See ../reserved-memory/reserved-memory.txt for details.
+
+ mediatek,topckgen:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the mediatek topckgen controller
+
+ power-domains:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: 26M clock
+ - description: audio pll1 clock
+ - description: audio pll2 clock
+ - description: clock divider for i2si1_mck
+ - description: clock divider for i2si2_mck
+ - description: clock divider for i2so1_mck
+ - description: clock divider for i2so2_mck
+ - description: clock divider for dptx_mck
+ - description: a1sys hoping clock
+ - description: audio intbus clock
+ - description: audio hires clock
+ - description: audio local bus clock
+ - description: mux for dptx_mck
+ - description: mux for i2so1_mck
+ - description: mux for i2so2_mck
+ - description: mux for i2si1_mck
+ - description: mux for i2si2_mck
+ - description: audio infra 26M clock
+ - description: infra bus clock
+
+ clock-names:
+ items:
+ - const: clk26m
+ - const: apll1_ck
+ - const: apll2_ck
+ - const: apll12_div0
+ - const: apll12_div1
+ - const: apll12_div2
+ - const: apll12_div3
+ - const: apll12_div9
+ - const: a1sys_hp_sel
+ - const: aud_intbus_sel
+ - const: audio_h_sel
+ - const: audio_local_bus_sel
+ - const: dptx_m_sel
+ - const: i2so1_m_sel
+ - const: i2so2_m_sel
+ - const: i2si1_m_sel
+ - const: i2si2_m_sel
+ - const: infra_ao_audio_26m_b
+ - const: scp_adsp_audiodsp
+
+ mediatek,etdm-in1-chn-disabled:
+ $ref: /schemas/types.yaml#/definitions/uint8-array
+ maxItems: 24
+ description: Specify which input channel should be disabled.
+
+ mediatek,etdm-in2-chn-disabled:
+ $ref: /schemas/types.yaml#/definitions/uint8-array
+ maxItems: 16
+ description: Specify which input channel should be disabled.
+
+patternProperties:
+ "^mediatek,etdm-in[1-2]-mclk-always-on-rate-hz$":
+ description: Specify etdm in mclk output rate for always on case.
+
+ "^mediatek,etdm-out[1-3]-mclk-always-on-rate-hz$":
+ description: Specify etdm out mclk output rate for always on case.
+
+ "^mediatek,etdm-in[1-2]-multi-pin-mode$":
+ type: boolean
+ description: if present, the etdm data mode is I2S.
+
+ "^mediatek,etdm-out[1-3]-multi-pin-mode$":
+ type: boolean
+ description: if present, the etdm data mode is I2S.
+
+ "^mediatek,etdm-in[1-2]-cowork-source$":
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: |
+ etdm modules can share the same external clock pin. Specify
+ which etdm clock source is required by this etdm in module.
+ enum:
+ - 0 # etdm1_in
+ - 1 # etdm2_in
+ - 2 # etdm1_out
+ - 3 # etdm2_out
+
+ "^mediatek,etdm-out[1-2]-cowork-source$":
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: |
+ etdm modules can share the same external clock pin. Specify
+ which etdm clock source is required by this etdm out module.
+ enum:
+ - 0 # etdm1_in
+ - 1 # etdm2_in
+ - 2 # etdm1_out
+ - 3 # etdm2_out
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - resets
+ - reset-names
+ - mediatek,topckgen
+ - power-domains
+ - clocks
+ - clock-names
+ - memory-region
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ afe: mt8195-afe-pcm@10890000 {
+ compatible = "mediatek,mt8195-audio";
+ reg = <0x10890000 0x10000>;
+ interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>;
+ resets = <&watchdog 14>;
+ reset-names = "audiosys";
+ mediatek,topckgen = <&topckgen>;
+ power-domains = <&spm 7>; //MT8195_POWER_DOMAIN_AUDIO
+ memory-region = <&snd_dma_mem_reserved>;
+ clocks = <&clk26m>,
+ <&topckgen 163>, //CLK_TOP_APLL1
+ <&topckgen 166>, //CLK_TOP_APLL2
+ <&topckgen 233>, //CLK_TOP_APLL12_DIV0
+ <&topckgen 234>, //CLK_TOP_APLL12_DIV1
+ <&topckgen 235>, //CLK_TOP_APLL12_DIV2
+ <&topckgen 236>, //CLK_TOP_APLL12_DIV3
+ <&topckgen 238>, //CLK_TOP_APLL12_DIV9
+ <&topckgen 100>, //CLK_TOP_A1SYS_HP_SEL
+ <&topckgen 33>, //CLK_TOP_AUD_INTBUS_SEL
+ <&topckgen 34>, //CLK_TOP_AUDIO_H_SEL
+ <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS_SEL
+ <&topckgen 98>, //CLK_TOP_DPTX_M_SEL
+ <&topckgen 94>, //CLK_TOP_I2SO1_M_SEL
+ <&topckgen 95>, //CLK_TOP_I2SO2_M_SEL
+ <&topckgen 96>, //CLK_TOP_I2SI1_M_SEL
+ <&topckgen 97>, //CLK_TOP_I2SI2_M_SEL
+ <&infracfg_ao 50>, //CLK_INFRA_AO_AUDIO_26M_B
+ <&scp_adsp 0>; //CLK_SCP_ADSP_AUDIODSP
+ clock-names = "clk26m",
+ "apll1_ck",
+ "apll2_ck",
+ "apll12_div0",
+ "apll12_div1",
+ "apll12_div2",
+ "apll12_div3",
+ "apll12_div9",
+ "a1sys_hp_sel",
+ "aud_intbus_sel",
+ "audio_h_sel",
+ "audio_local_bus_sel",
+ "dptx_m_sel",
+ "i2so1_m_sel",
+ "i2so2_m_sel",
+ "i2si1_m_sel",
+ "i2si2_m_sel",
+ "infra_ao_audio_26m_b",
+ "scp_adsp_audiodsp";
+ };
+
+...
diff --git a/Bindings/sound/mt8195-mt6359.yaml b/Bindings/sound/mt8195-mt6359.yaml
new file mode 100644
index 0000000..c1ddbf6
--- /dev/null
+++ b/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/Bindings/sound/mtk-afe-pcm.txt b/Bindings/sound/mtk-afe-pcm.txt
new file mode 100644
index 0000000..e302c7f
--- /dev/null
+++ b/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/Bindings/sound/mtk-btcvsd-snd.txt b/Bindings/sound/mtk-btcvsd-snd.txt
new file mode 100644
index 0000000..679e448
--- /dev/null
+++ b/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/Bindings/sound/mvebu-audio.txt b/Bindings/sound/mvebu-audio.txt
new file mode 100644
index 0000000..4f5dec5
--- /dev/null
+++ b/Bindings/sound/mvebu-audio.txt
@@ -0,0 +1,46 @@
+* 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
+ "marvell,armada-380-audio" for Armada 38x platforms
+
+- reg: physical base address of the controller and length of memory mapped
+ region (named "i2s_regs").
+ With "marvell,armada-380-audio" two other regions are required:
+ first of those is dedicated for Audio PLL Configuration registers
+ (named "pll_regs") and the second one ("soc_ctrl") - for register
+ where one of exceptive I/O types (I2S or S/PDIF) is set.
+
+- interrupts:
+ with "marvell,kirkwood-audio", the audio interrupt
+ 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
+
+Optional properties:
+
+- spdif-mode:
+ Enable S/PDIF mode on Armada 38x SoC. Using this property
+ disables standard I2S I/O. Valid only with "marvell,armada-380-audio"
+ compatible string.
+
+Example:
+
+i2s1: audio-controller@b4000 {
+ compatible = "marvell,dove-audio";
+ reg = <0xb4000 0x2210>;
+ interrupts = <21>, <22>;
+ clocks = <&gate_clk 13>;
+ clock-names = "internal";
+};
diff --git a/Bindings/sound/mxs-audio-sgtl5000.txt b/Bindings/sound/mxs-audio-sgtl5000.txt
new file mode 100644
index 0000000..4eb980b
--- /dev/null
+++ b/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/Bindings/sound/mxs-saif.txt b/Bindings/sound/mxs-saif.txt
new file mode 100644
index 0000000..7ba07a1
--- /dev/null
+++ b/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/Bindings/sound/nokia,rx51.txt b/Bindings/sound/nokia,rx51.txt
new file mode 100644
index 0000000..72f93d9
--- /dev/null
+++ b/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/Bindings/sound/nuvoton,nau8315.yaml b/Bindings/sound/nuvoton,nau8315.yaml
new file mode 100644
index 0000000..24006e9
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8315.yaml
@@ -0,0 +1,44 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8315.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8315/NAU8318 Mono Class-D Amplifier
+
+maintainers:
+ - David Lin <CTLIN0@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - nuvoton,nau8315
+ - nuvoton,nau8318
+
+ '#sound-dai-cells':
+ const: 0
+
+ enable-gpios:
+ maxItems: 1
+ description:
+ GPIO specifier for the chip's device enable input(EN) pin.
+ If this option is not specified then driver does not manage
+ the pin state (e.g. chip is always on).
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ codec {
+ compatible = "nuvoton,nau8315";
+ #sound-dai-cells = <0>;
+ enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>;
+ };
diff --git a/Bindings/sound/nuvoton,nau8540.yaml b/Bindings/sound/nuvoton,nau8540.yaml
new file mode 100644
index 0000000..7ccfbb8
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8540.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8540.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Nuvoton Technology Corporation NAU85L40 Audio CODEC
+
+maintainers:
+ - John Hsu <KCHSU0@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: nuvoton,nau8540
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1c {
+ compatible = "nuvoton,nau8540";
+ reg = <0x1c>;
+ };
+ };
diff --git a/Bindings/sound/nuvoton,nau8810.yaml b/Bindings/sound/nuvoton,nau8810.yaml
new file mode 100644
index 0000000..d9696f6
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8810.yaml
@@ -0,0 +1,45 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8810.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8810/NAU8812/NAU8814 audio CODEC
+
+maintainers:
+ - David Lin <CTLIN0@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - nuvoton,nau8810
+ - nuvoton,nau8812
+ - nuvoton,nau8814
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ #sound-dai-cells = <0>;
+ compatible = "nuvoton,nau8810";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/nuvoton,nau8821.yaml b/Bindings/sound/nuvoton,nau8821.yaml
new file mode 100644
index 0000000..3e54abd
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8821.yaml
@@ -0,0 +1,132 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU88L21 audio codec
+
+maintainers:
+ - Seven Lee <wtli@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: nuvoton,nau8821
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ nuvoton,jkdet-enable:
+ description: Enable jack detection via JKDET pin.
+ type: boolean
+
+ nuvoton,jkdet-pull-enable:
+ description: Enable JKDET pin pull. If set - pin pull enabled,
+ otherwise pin in high impedance state.
+ type: boolean
+
+ nuvoton,jkdet-pull-up:
+ description: Pull-up JKDET pin. If set then JKDET pin is pull up,
+ otherwise pull down.
+ type: boolean
+
+ nuvoton,key-enable:
+ description: handles key press detection.
+ type: boolean
+
+ nuvoton,jkdet-polarity:
+ description: JKDET pin polarity.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # active high
+ - 1 # active low
+ default: 1
+
+ nuvoton,micbias-voltage:
+ description: MICBIAS output level select.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # VDDA
+ - 1 # VDDA * 1
+ - 2 # VDDA * 1.1
+ - 3 # VDDA * 1.2
+ - 4 # VDDA * 1.3
+ - 5 # VDDA * 1.4
+ - 6 # VDDA * 1.53
+ - 7 # VDDA * 1.53
+ default: 6
+
+ nuvoton,vref-impedance:
+ description: VMID Tie-off impedance select.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # open
+ - 1 # 25KOhms
+ - 2 # 125KOhms
+ - 3 # 2.5KOhms
+ default: 2
+
+ nuvoton,jack-insert-debounce:
+ description: number from 0 to 7 that sets debounce time to 2^(n+2)ms.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ maximum: 7
+ default: 7
+
+ nuvoton,jack-eject-debounce:
+ description: number from 0 to 7 that sets debounce time to 2^(n+2)ms.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ maximum: 7
+ default: 0
+
+ nuvoton,dmic-clk-threshold:
+ description: DMIC clock speed expected value. Unit is Hz.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ default: 3072000
+
+ nuvoton,left-input-single-end:
+ description: Enable left input with single-ended settings if set.
+ For the headset mic application, the single-ended control is
+ just limited to the left adc for design demand.
+ type: boolean
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1b {
+ compatible = "nuvoton,nau8821";
+ reg = <0x1b>;
+ interrupt-parent = <&gpio>;
+ interrupts = <23 IRQ_TYPE_LEVEL_LOW>;
+ nuvoton,jkdet-enable;
+ nuvoton,jkdet-pull-enable;
+ nuvoton,jkdet-pull-up;
+ nuvoton,key-enable;
+ nuvoton,left-input-single-end;
+ nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>;
+ nuvoton,micbias-voltage = <6>;
+ nuvoton,vref-impedance = <2>;
+ nuvoton,jack-insert-debounce = <7>;
+ nuvoton,jack-eject-debounce = <0>;
+ nuvoton,dmic-clk-threshold = <3072000>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/nuvoton,nau8822.yaml b/Bindings/sound/nuvoton,nau8822.yaml
new file mode 100644
index 0000000..cb8182b
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8822.yaml
@@ -0,0 +1,58 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8822.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8822 audio CODEC
+
+description: |
+ 24 bit stereo audio codec with speaker driver.
+ This device supports I2C/SPI.
+
+maintainers:
+ - David Lin <CTLIN0@nuvoton.com>
+
+properties:
+ compatible:
+ enum:
+ - nuvoton,nau8822
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: mclk
+
+ nuvoton,spk-btl:
+ description:
+ If set, configure the two loudspeaker outputs as a Bridge Tied Load output
+ to drive a high power external loudspeaker.
+ $ref: /schemas/types.yaml#/definitions/flag
+
+required:
+ - compatible
+ - reg
+
+allOf:
+ - $ref: dai-common.yaml#
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "nuvoton,nau8822";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/nuvoton,nau8824.yaml b/Bindings/sound/nuvoton,nau8824.yaml
new file mode 100644
index 0000000..3dbf438
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8824.yaml
@@ -0,0 +1,182 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8824.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8824 audio CODEC
+
+maintainers:
+ - John Hsu <KCHSU0@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - nuvoton,nau8824
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ interrupts:
+ maxItems: 1
+
+ nuvoton,jkdet-polarity:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ JKDET pin polarity.
+ enum:
+ - 0 # active high
+ - 1 # active low
+ default: 1
+
+ nuvoton,vref-impedance:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ VREF Impedance selection.
+ enum:
+ - 0 # Open
+ - 1 # 25 kOhm
+ - 2 # 125 kOhm
+ - 3 # 2.5 kOhm
+ default: 2
+
+ nuvoton,micbias-voltage:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Micbias voltage level.
+ enum:
+ - 0 # VDDA
+ - 1 # VDDA
+ - 2 # VDDA * 1.1
+ - 3 # VDDA * 1.2
+ - 4 # VDDA * 1.3
+ - 5 # VDDA * 1.4
+ - 6 # VDDA * 1.53
+ - 7 # VDDA * 1.53
+ default: 6
+
+ nuvoton,sar-threshold-num:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Number of buttons supported.
+ minimum: 1
+ maximum: 8
+ default: 4
+
+ nuvoton,sar-threshold:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Impedance threshold for each button. Array that contains up to 8 buttons
+ configuration. SAR value is calculated as
+ SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is
+ configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by
+ 'nuvoton,sar-voltage', R - button impedance.
+ Refer datasheet section 10.2 for more information about threshold
+ calculation.
+ minItems: 1
+ maxItems: 8
+ items:
+ minimum: 0
+ maximum: 255
+
+ nuvoton,sar-hysteresis:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Button impedance measurement hysteresis.
+ default: 0
+
+ nuvoton,sar-voltage:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Reference voltage for button impedance measurement.
+ enum:
+ - 0 # VDDA
+ - 1 # VDDA
+ - 2 # VDDA * 1.1
+ - 3 # VDDA * 1.2
+ - 4 # VDDA * 1.3
+ - 5 # VDDA * 1.4
+ - 6 # VDDA * 1.53
+ - 7 # VDDA * 1.53
+ default: 6
+
+ nuvoton,sar-compare-time:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ SAR compare time.
+ enum:
+ - 0 # 500ns
+ - 1 # 1us
+ - 2 # 2us
+ - 3 # 4us
+ default: 1
+
+ nuvoton,sar-sampling-time:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ SAR sampling time.
+ enum:
+ - 0 # 2us
+ - 1 # 4us
+ - 2 # 8us
+ - 3 # 16us
+ default: 1
+
+ nuvoton,short-key-debounce:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Button short key press debounce time.
+ enum:
+ - 0 # 30 ms
+ - 1 # 50 ms
+ - 2 # 100 ms
+ default: 0
+
+ nuvoton,jack-eject-debounce:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Jack ejection debounce time.
+ enum:
+ - 0 # 0 ms
+ - 1 # 1 ms
+ - 2 # 10 ms
+ default: 1
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ #sound-dai-cells = <0>;
+ compatible = "nuvoton,nau8824";
+ reg = <0x1a>;
+ interrupt-parent = <&gpio>;
+ interrupts = <38 IRQ_TYPE_LEVEL_LOW>;
+ nuvoton,vref-impedance = <2>;
+ nuvoton,micbias-voltage = <6>;
+ nuvoton,sar-threshold-num = <4>;
+ // Setup 4 buttons impedance according to Android specification
+ nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+ nuvoton,sar-hysteresis = <0>;
+ nuvoton,sar-voltage = <6>;
+ nuvoton,sar-compare-time = <1>;
+ nuvoton,sar-sampling-time = <1>;
+ nuvoton,short-key-debounce = <0>;
+ nuvoton,jack-eject-debounce = <1>;
+ };
+ };
diff --git a/Bindings/sound/nuvoton,nau8825.yaml b/Bindings/sound/nuvoton,nau8825.yaml
new file mode 100644
index 0000000..a54f194
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8825.yaml
@@ -0,0 +1,239 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8825.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8825 audio CODEC
+
+maintainers:
+ - John Hsu <KCHSU0@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - nuvoton,nau8825
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ nuvoton,jkdet-enable:
+ description:
+ Enable jack detection via JKDET pin.
+ type: boolean
+
+ nuvoton,jkdet-pull-enable:
+ description:
+ Enable JKDET pin pull.
+ If set - pin pull enabled, otherwise pin in high impedance state.
+ type: boolean
+
+ nuvoton,jkdet-pull-up:
+ description:
+ Pull-up JKDET pin.
+ If set then JKDET pin is pull up, otherwise pull down.
+ type: boolean
+
+ nuvoton,jkdet-polarity:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ JKDET pin polarity.
+ enum:
+ - 0 # active high
+ - 1 # active low
+ default: 1
+
+ nuvoton,vref-impedance:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ VREF Impedance selection.
+ enum:
+ - 0 # Open
+ - 1 # 25 kOhm
+ - 2 # 125 kOhm
+ - 3 # 2.5 kOhm
+ default: 2
+
+ nuvoton,micbias-voltage:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Micbias voltage level.
+ enum:
+ - 0 # VDDA
+ - 1 # VDDA
+ - 2 # VDDA * 1.1
+ - 3 # VDDA * 1.2
+ - 4 # VDDA * 1.3
+ - 5 # VDDA * 1.4
+ - 6 # VDDA * 1.53
+ - 7 # VDDA * 1.53
+ default: 6
+
+ nuvoton,sar-threshold-num:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Number of buttons supported.
+ minimum: 1
+ maximum: 4
+ default: 4
+
+ nuvoton,sar-threshold:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Impedance threshold for each button. Array that contains up to 8 buttons
+ configuration. SAR value is calculated as
+ SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is
+ configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by
+ 'nuvoton,sar-voltage', R - button impedance.
+ Refer datasheet section 10.2 for more information about threshold
+ calculation.
+ minItems: 1
+ maxItems: 4
+ items:
+ minimum: 0
+ maximum: 255
+
+ nuvoton,sar-hysteresis:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Button impedance measurement hysteresis.
+ default: 0
+
+ nuvoton,sar-voltage:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Reference voltage for button impedance measurement.
+ enum:
+ - 0 # VDDA
+ - 1 # VDDA
+ - 2 # VDDA * 1.1
+ - 3 # VDDA * 1.2
+ - 4 # VDDA * 1.3
+ - 5 # VDDA * 1.4
+ - 6 # VDDA * 1.53
+ - 7 # VDDA * 1.53
+ default: 6
+
+ nuvoton,sar-compare-time:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ SAR compare time.
+ enum:
+ - 0 # 500 ns
+ - 1 # 1 us
+ - 2 # 2 us
+ - 3 # 4 us
+ default: 1
+
+ nuvoton,sar-sampling-time:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ SAR sampling time.
+ enum:
+ - 0 # 2 us
+ - 1 # 4 us
+ - 2 # 8 us
+ - 3 # 16 us
+ default: 1
+
+ nuvoton,short-key-debounce:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Button short key press debounce time.
+ enum:
+ - 0 # 30 ms
+ - 1 # 50 ms
+ - 2 # 100 ms
+ - 3 # 30 ms
+ default: 3
+
+ nuvoton,jack-insert-debounce:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ number from 0 to 7 that sets debounce time to 2^(n+2) ms.
+ maximum: 7
+ default: 7
+
+ nuvoton,jack-eject-debounce:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ number from 0 to 7 that sets debounce time to 2^(n+2) ms
+ maximum: 7
+ default: 0
+
+ nuvoton,crosstalk-enable:
+ description:
+ make crosstalk function enable if set.
+ type: boolean
+
+ nuvoton,adcout-drive-strong:
+ description:
+ make the drive strength of ADCOUT IO PIN strong if set.
+ Otherwise, the drive keeps normal strength.
+ type: boolean
+
+ nuvoton,adc-delay-ms:
+ description:
+ Delay (in ms) to make input path stable and avoid pop noise.
+ The default value is 125 and range between 125 to 500 ms.
+ minimum: 125
+ maximum: 500
+ default: 125
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: mclk
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ #sound-dai-cells = <0>;
+ compatible = "nuvoton,nau8825";
+ reg = <0x1a>;
+ interrupt-parent = <&gpio>;
+ interrupts = <38 IRQ_TYPE_LEVEL_LOW>;
+ nuvoton,jkdet-enable;
+ nuvoton,jkdet-pull-enable;
+ nuvoton,jkdet-pull-up;
+ nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>;
+ nuvoton,vref-impedance = <2>;
+ nuvoton,micbias-voltage = <6>;
+ // Setup 4 buttons impedance according to Android specification
+ nuvoton,sar-threshold-num = <4>;
+ nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+ nuvoton,sar-hysteresis = <1>;
+ nuvoton,sar-voltage = <0>;
+ nuvoton,sar-compare-time = <0>;
+ nuvoton,sar-sampling-time = <0>;
+ nuvoton,short-key-debounce = <2>;
+ nuvoton,jack-insert-debounce = <7>;
+ nuvoton,jack-eject-debounce = <7>;
+ nuvoton,crosstalk-enable;
+
+ clock-names = "mclk";
+ clocks = <&tegra_pmc 1>;
+ };
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-alc5632.yaml b/Bindings/sound/nvidia,tegra-audio-alc5632.yaml
new file mode 100644
index 0000000..96f2f92
--- /dev/null
+++ b/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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headset Stereophone
+ - Int Spk
+ - Headset Mic
+ - Digital Mic
+
+ # CODEC Pins
+ - SPKOUT
+ - SPKOUTN
+ - MICBIAS1
+ - MIC1
+ - HPR
+ - HPL
+ - DMICDAT
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-alc5632-paz00",
+ "nvidia,tegra-audio-alc5632";
+
+ nvidia,model = "Compal PAZ00";
+
+ nvidia,audio-routing = "Int Spk", "SPKOUT",
+ "Int Spk", "SPKOUTN",
+ "Headset Mic", "MICBIAS1",
+ "MIC1", "Headset Mic",
+ "Headset Stereophone", "HPR",
+ "Headset Stereophone", "HPL",
+ "DMICDAT", "Digital Mic";
+
+ nvidia,i2s-controller = <&i2s>;
+ nvidia,audio-codec = <&codec>;
+
+ clocks = <&clk 112>, <&clk 113>, <&clk 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-common.yaml b/Bindings/sound/nvidia,tegra-audio-common.yaml
new file mode 100644
index 0000000..2588589
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-common.yaml
@@ -0,0 +1,87 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-common.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Common properties for NVIDIA Tegra audio complexes
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Thierry Reding <thierry.reding@gmail.com>
+
+properties:
+ clocks:
+ items:
+ - description: PLL A clock
+ - description: PLL A OUT0 clock
+ - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk
+
+ clock-names:
+ items:
+ - const: pll_a
+ - const: pll_a_out0
+ - const: mclk
+
+ nvidia,model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: The user-visible name of this sound complex.
+
+ nvidia,audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+
+ nvidia,ac97-controller:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the AC97 controller
+
+ nvidia,i2s-controller:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the Tegra I2S controller
+
+ nvidia,audio-codec:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of audio codec
+
+ nvidia,spkr-en-gpios:
+ maxItems: 1
+ description: The GPIO that enables the speakers
+
+ nvidia,hp-mute-gpios:
+ maxItems: 1
+ description: The GPIO that mutes the headphones
+
+ nvidia,hp-det-gpios:
+ maxItems: 1
+ description: The GPIO that detect headphones are plugged in
+
+ nvidia,mic-det-gpios:
+ maxItems: 1
+ description: The GPIO that detect microphone is plugged in
+
+ nvidia,ear-sel-gpios:
+ maxItems: 1
+ description: The GPIO that switch between the microphones
+
+ nvidia,int-mic-en-gpios:
+ maxItems: 1
+ description: The GPIO that enables the internal microphone
+
+ nvidia,ext-mic-en-gpios:
+ maxItems: 1
+ description: The GPIO that enables the external microphone
+
+ nvidia,headset:
+ type: boolean
+ description: The Mic Jack represents state of the headset microphone pin
+
+ nvidia,coupled-mic-hp-det:
+ type: boolean
+ description: The Mic detect GPIO is viable only if HP detect GPIO is active
+
+additionalProperties: true
diff --git a/Bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Bindings/sound/nvidia,tegra-audio-graph-card.yaml
new file mode 100644
index 0000000..b4bee46
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-graph-card.yaml
@@ -0,0 +1,199 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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/Bindings/sound/nvidia,tegra-audio-max9808x.yaml b/Bindings/sound/nvidia,tegra-audio-max9808x.yaml
new file mode 100644
index 0000000..c29d794
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-max9808x.yaml
@@ -0,0 +1,90 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-max9808x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra audio complex with MAX9808x CODEC
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Thierry Reding <thierry.reding@gmail.com>
+
+allOf:
+ - $ref: nvidia,tegra-audio-common.yaml#
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - pattern: '^[a-z0-9]+,tegra-audio-max98088(-[a-z0-9]+)+$'
+ - const: nvidia,tegra-audio-max98088
+ - items:
+ - pattern: '^[a-z0-9]+,tegra-audio-max98089(-[a-z0-9]+)+$'
+ - const: nvidia,tegra-audio-max98089
+
+ nvidia,audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Int Spk
+ - Headphone Jack
+ - Earpiece
+ - Headset Mic
+ - Internal Mic 1
+ - Internal Mic 2
+
+ # CODEC Pins
+ - HPL
+ - HPR
+ - SPKL
+ - SPKR
+ - RECL
+ - RECR
+ - INA1
+ - INA2
+ - INB1
+ - INB2
+ - MIC1
+ - MIC2
+ - MICBIAS
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/tegra30-car.h>
+ #include <dt-bindings/soc/tegra-pmc.h>
+ sound {
+ compatible = "lge,tegra-audio-max98089-p895",
+ "nvidia,tegra-audio-max98089";
+ nvidia,model = "LG Optimus Vu MAX98089";
+
+ nvidia,audio-routing =
+ "Headphone Jack", "HPL",
+ "Headphone Jack", "HPR",
+ "Int Spk", "SPKL",
+ "Int Spk", "SPKR",
+ "Earpiece", "RECL",
+ "Earpiece", "RECR",
+ "INA1", "Headset Mic",
+ "MIC1", "MICBIAS",
+ "MICBIAS", "Internal Mic 1",
+ "MIC2", "Internal Mic 2";
+
+ nvidia,i2s-controller = <&tegra_i2s0>;
+ nvidia,audio-codec = <&codec>;
+
+ clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
+ <&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
+ <&tegra_pmc TEGRA_PMC_CLK_OUT_1>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-max98090.yaml b/Bindings/sound/nvidia,tegra-audio-max98090.yaml
new file mode 100644
index 0000000..4d91245
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-max98090.yaml
@@ -0,0 +1,97 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphones
+ - Speakers
+ - Mic Jack
+ - Int Mic
+
+ # CODEC Pins
+ - MIC1
+ - MIC2
+ - DMICL
+ - DMICR
+ - IN1
+ - IN2
+ - IN3
+ - IN4
+ - IN5
+ - IN6
+ - IN12
+ - IN34
+ - IN56
+ - HPL
+ - HPR
+ - SPKL
+ - SPKR
+ - RCVL
+ - RCVR
+ - MICBIAS
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/tegra124-car.h>
+
+ sound {
+ compatible = "nvidia,tegra-audio-max98090-venice2",
+ "nvidia,tegra-audio-max98090";
+ nvidia,model = "NVIDIA Tegra Venice2";
+
+ nvidia,audio-routing =
+ "Headphones", "HPR",
+ "Headphones", "HPL",
+ "Speakers", "SPKR",
+ "Speakers", "SPKL",
+ "Mic Jack", "MICBIAS",
+ "IN34", "Mic Jack";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&acodec>;
+
+ clocks = <&tegra_car TEGRA124_CLK_PLL_A>,
+ <&tegra_car TEGRA124_CLK_PLL_A_OUT0>,
+ <&tegra_car TEGRA124_CLK_EXTERN1>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-rt5631.yaml b/Bindings/sound/nvidia,tegra-audio-rt5631.yaml
new file mode 100644
index 0000000..0c8067c
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-rt5631.yaml
@@ -0,0 +1,85 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5631.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra audio complex with RT5631 CODEC
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Thierry Reding <thierry.reding@gmail.com>
+
+allOf:
+ - $ref: nvidia,tegra-audio-common.yaml#
+
+properties:
+ compatible:
+ items:
+ - pattern: '^[a-z0-9]+,tegra-audio-rt5631(-[a-z0-9]+)+$'
+ - const: nvidia,tegra-audio-rt5631
+
+ nvidia,audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Int Spk
+ - Headphone Jack
+ - Mic Jack
+ - Int Mic
+
+ # CODEC Pins
+ - MIC1
+ - MIC2
+ - AXIL
+ - AXIR
+ - MONOIN_RXN
+ - MONOIN_RXP
+ - DMIC
+ - MIC Bias1
+ - MIC Bias2
+ - MONO_IN
+ - AUXO1
+ - AUXO2
+ - SPOL
+ - SPOR
+ - HPOL
+ - HPOR
+ - MONO
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/tegra30-car.h>
+ #include <dt-bindings/soc/tegra-pmc.h>
+ sound {
+ compatible = "asus,tegra-audio-rt5631-tf700t",
+ "nvidia,tegra-audio-rt5631";
+ nvidia,model = "Asus Transformer Infinity TF700T RT5631";
+
+ nvidia,audio-routing =
+ "Headphone Jack", "HPOL",
+ "Headphone Jack", "HPOR",
+ "Int Spk", "SPOL",
+ "Int Spk", "SPOR",
+ "MIC1", "MIC Bias1",
+ "MIC Bias1", "Mic Jack",
+ "DMIC", "Int Mic";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&rt5631>;
+
+ clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
+ <&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
+ <&tegra_pmc TEGRA_PMC_CLK_OUT_1>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-rt5640.yaml b/Bindings/sound/nvidia,tegra-audio-rt5640.yaml
new file mode 100644
index 0000000..2638592
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-rt5640.yaml
@@ -0,0 +1,84 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphones
+ - Speakers
+ - Mic Jack
+
+ # CODEC Pins
+ - DMIC1
+ - DMIC2
+ - MICBIAS1
+ - IN1P
+ - IN1R
+ - IN2P
+ - IN2R
+ - HPOL
+ - HPOR
+ - LOUTL
+ - LOUTR
+ - MONOP
+ - MONON
+ - SPOLP
+ - SPOLN
+ - SPORP
+ - SPORN
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-rt5640-dalmore",
+ "nvidia,tegra-audio-rt5640";
+ nvidia,model = "NVIDIA Tegra Dalmore";
+
+ nvidia,audio-routing =
+ "Headphones", "HPOR",
+ "Headphones", "HPOL",
+ "Speakers", "SPORP",
+ "Speakers", "SPORN",
+ "Speakers", "SPOLP",
+ "Speakers", "SPOLN";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&rt5640>;
+
+ nvidia,hp-det-gpios = <&gpio 143 0>;
+
+ clocks = <&clk 216>, <&clk 217>, <&clk 120>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-rt5677.yaml b/Bindings/sound/nvidia,tegra-audio-rt5677.yaml
new file mode 100644
index 0000000..09e1d0b
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-rt5677.yaml
@@ -0,0 +1,100 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphone
+ - Speaker
+ - Headset Mic
+ - Internal Mic 1
+ - Internal Mic 2
+
+ # CODEC Pins
+ - IN1P
+ - IN1N
+ - IN2P
+ - IN2N
+ - MICBIAS1
+ - DMIC1
+ - DMIC2
+ - DMIC3
+ - DMIC4
+ - DMIC L1
+ - DMIC L2
+ - DMIC L3
+ - DMIC L4
+ - DMIC R1
+ - DMIC R2
+ - DMIC R3
+ - DMIC R4
+ - LOUT1
+ - LOUT2
+ - LOUT3
+ - PDM1L
+ - PDM1R
+ - PDM2L
+ - PDM2R
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-rt5677-ryu",
+ "nvidia,tegra-audio-rt5677";
+ nvidia,model = "NVIDIA Tegra Ryu";
+
+ nvidia,audio-routing =
+ "Headphone", "LOUT2",
+ "Headphone", "LOUT1",
+ "Headset Mic", "MICBIAS1",
+ "IN1P", "Headset Mic",
+ "IN1N", "Headset Mic",
+ "DMIC L1", "Internal Mic 1",
+ "DMIC R1", "Internal Mic 1",
+ "DMIC L2", "Internal Mic 2",
+ "DMIC R2", "Internal Mic 2",
+ "Speaker", "PDM1L",
+ "Speaker", "PDM1R";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&rt5677>;
+
+ nvidia,hp-det-gpios = <&gpio 143 0>;
+
+ clocks = <&clk 216>,
+ <&clk 217>,
+ <&clk 121>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-sgtl5000.yaml b/Bindings/sound/nvidia,tegra-audio-sgtl5000.yaml
new file mode 100644
index 0000000..e5bc6a6
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-sgtl5000.yaml
@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphone Jack
+ - Line In Jack
+ - Mic Jack
+
+ # CODEC Pins
+ - HP_OUT
+ - LINE_OUT
+ - LINE_IN
+ - MIC_IN
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/tegra30-car.h>
+
+ sound {
+ compatible = "toradex,tegra-audio-sgtl5000-apalis_t30",
+ "nvidia,tegra-audio-sgtl5000";
+ nvidia,model = "Toradex Apalis T30 SGTL5000";
+ nvidia,audio-routing =
+ "Headphone Jack", "HP_OUT",
+ "LINE_IN", "Line In Jack",
+ "MIC_IN", "Mic Jack";
+ nvidia,i2s-controller = <&tegra_i2s2>;
+ nvidia,audio-codec = <&codec>;
+ clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
+ <&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
+ <&tegra_car TEGRA30_CLK_EXTERN1>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-trimslice.yaml b/Bindings/sound/nvidia,tegra-audio-trimslice.yaml
new file mode 100644
index 0000000..8c87cd1
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-trimslice.yaml
@@ -0,0 +1,33 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ - $ref: nvidia,tegra-audio-common.yaml#
+
+properties:
+ compatible:
+ const: nvidia,tegra-audio-trimslice
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-trimslice";
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&codec>;
+ clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-wm8753.yaml b/Bindings/sound/nvidia,tegra-audio-wm8753.yaml
new file mode 100644
index 0000000..3323d6a
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-wm8753.yaml
@@ -0,0 +1,79 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphone Jack
+ - Mic Jack
+
+ # CODEC Pins
+ - LOUT1
+ - LOUT2
+ - ROUT1
+ - ROUT2
+ - MONO1
+ - MONO2
+ - OUT3
+ - OUT4
+ - LINE1
+ - LINE2
+ - RXP
+ - RXN
+ - ACIN
+ - ACOP
+ - MIC1N
+ - MIC1
+ - MIC2N
+ - MIC2
+ - Mic Bias
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-wm8753-whistler",
+ "nvidia,tegra-audio-wm8753";
+ nvidia,model = "tegra-wm8753-harmony";
+
+ nvidia,audio-routing =
+ "Headphone Jack", "LOUT1",
+ "Headphone Jack", "ROUT1";
+
+ nvidia,i2s-controller = <&i2s1>;
+ nvidia,audio-codec = <&wm8753>;
+
+ clocks = <&clk 112>, <&clk 113>, <&clk 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-wm8903.yaml b/Bindings/sound/nvidia,tegra-audio-wm8903.yaml
new file mode 100644
index 0000000..1be25ce
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-wm8903.yaml
@@ -0,0 +1,93 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphone Jack
+ - Int Spk
+ - Mic Jack
+ - Int Mic
+
+ # CODEC Pins
+ - IN1L
+ - IN1R
+ - IN2L
+ - IN2R
+ - IN3L
+ - IN3R
+ - DMICDAT
+ - HPOUTL
+ - HPOUTR
+ - LINEOUTL
+ - LINEOUTR
+ - LOP
+ - LON
+ - ROP
+ - RON
+ - MICBIAS
+
+required:
+ - nvidia,i2s-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-wm8903-harmony",
+ "nvidia,tegra-audio-wm8903";
+ nvidia,model = "tegra-wm8903-harmony";
+
+ nvidia,audio-routing =
+ "Headphone Jack", "HPOUTR",
+ "Headphone Jack", "HPOUTL",
+ "Int Spk", "ROP",
+ "Int Spk", "RON",
+ "Int Spk", "LOP",
+ "Int Spk", "LON",
+ "Mic Jack", "MICBIAS",
+ "IN1L", "Mic Jack";
+
+ nvidia,i2s-controller = <&i2s1>;
+ nvidia,audio-codec = <&wm8903>;
+
+ nvidia,spkr-en-gpios = <&codec 2 0>;
+ nvidia,hp-det-gpios = <&gpio 178 0>;
+ nvidia,int-mic-en-gpios = <&gpio 184 0>;
+ nvidia,ext-mic-en-gpios = <&gpio 185 0>;
+
+ clocks = <&clk 112>, <&clk 113>, <&clk 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra-audio-wm9712.yaml b/Bindings/sound/nvidia,tegra-audio-wm9712.yaml
new file mode 100644
index 0000000..397306b
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra-audio-wm9712.yaml
@@ -0,0 +1,76 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source. Valid names for sources and
+ sinks are the pins (documented in the binding document),
+ and the jacks on the board.
+ minItems: 2
+ items:
+ enum:
+ # Board Connectors
+ - Headphone
+ - LineIn
+ - Mic
+
+ # CODEC Pins
+ - MONOOUT
+ - HPOUTL
+ - HPOUTR
+ - LOUT2
+ - ROUT2
+ - OUT3
+ - LINEINL
+ - LINEINR
+ - PHONE
+ - PCBEEP
+ - MIC1
+ - MIC2
+ - Mic Bias
+
+required:
+ - nvidia,ac97-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "nvidia,tegra-audio-wm9712-colibri_t20",
+ "nvidia,tegra-audio-wm9712";
+ nvidia,model = "Toradex Colibri T20";
+
+ nvidia,audio-routing =
+ "Headphone", "HPOUTL",
+ "Headphone", "HPOUTR",
+ "LineIn", "LINEINL",
+ "LineIn", "LINEINR",
+ "Mic", "MIC1";
+
+ nvidia,ac97-controller = <&ac97>;
+
+ clocks = <&clk 112>, <&clk 113>, <&clk 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+ };
diff --git a/Bindings/sound/nvidia,tegra186-asrc.yaml b/Bindings/sound/nvidia,tegra186-asrc.yaml
new file mode 100644
index 0000000..e15f387
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra186-asrc.yaml
@@ -0,0 +1,81 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra186-asrc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra186 ASRC
+
+description: |
+ Asynchronous Sample Rate Converter (ASRC) converts the sampling frequency
+ of the input signal from one frequency to another. It can handle over a
+ wide range of sample rate ratios (freq_in/freq_out) from 1:24 to 24:1.
+ ASRC has two modes of operation. One where ratio can be programmed in SW
+ and the other where it gets the information from ratio estimator module.
+
+ It supports sample rate conversions in the range of 8 to 192 kHz and
+ supports 6 streams upto 12 total channels. The input data size can be
+ 16, 24 and 32 bits.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^asrc@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra186-asrc
+ - items:
+ - enum:
+ - nvidia,tegra234-asrc
+ - nvidia,tegra194-asrc
+ - const: nvidia,tegra186-asrc
+
+ reg:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^ASRC[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ description: |
+ ASRC has seven input ports and six output ports. Accordingly ACIF
+ (Audio Client Interfaces) port nodes are defined to represent the
+ ASRC inputs (port 0 to 6) and outputs (port 7 to 12). These are
+ connected to corresponding ports on AHUB (Audio Hub). Additional
+ input (port 6) is for receiving ratio information from estimator.
+
+ patternProperties:
+ '^port@[0-6]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: ASRC ACIF input ports
+ '^port@[7-9]|1[1-2]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: ASRC ACIF output ports
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+
+ asrc@2910000 {
+ compatible = "nvidia,tegra186-asrc";
+ reg = <0x2910000 0x2000>;
+ sound-name-prefix = "ASRC1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra186-dspk.yaml b/Bindings/sound/nvidia,tegra186-dspk.yaml
new file mode 100644
index 0000000..e1362c7
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra186-dspk.yaml
@@ -0,0 +1,100 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra186-dspk.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra186 DSPK Controller
+
+description: |
+ The Digital Speaker Controller (DSPK) can be viewed as a Pulse
+ Density Modulation (PDM) transmitter that up-samples the input to
+ the desired sampling rate by interpolation and then converts the
+ over sampled Pulse Code Modulation (PCM) input to the desired 1-bit
+ output via Delta Sigma Modulation (DSM).
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^dspk@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra186-dspk
+ - items:
+ - enum:
+ - nvidia,tegra234-dspk
+ - nvidia,tegra194-dspk
+ - const: nvidia,tegra186-dspk
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: dspk
+
+ assigned-clocks:
+ maxItems: 1
+
+ assigned-clock-parents:
+ maxItems: 1
+
+ assigned-clock-rates:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^DSPK[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ DSPK ACIF (Audio Client Interface) port connected to the
+ corresponding AHUB (Audio Hub) ACIF port.
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ DSPK DAP (Digital Audio Port) interface which can be connected
+ to external audio codec for playback.
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - assigned-clocks
+ - assigned-clock-parents
+ - sound-name-prefix
+
+additionalProperties: false
+
+examples:
+ - |
+ #include<dt-bindings/clock/tegra186-clock.h>
+
+ dspk@2905000 {
+ compatible = "nvidia,tegra186-dspk";
+ reg = <0x2905000 0x100>;
+ clocks = <&bpmp TEGRA186_CLK_DSPK1>;
+ clock-names = "dspk";
+ assigned-clocks = <&bpmp TEGRA186_CLK_DSPK1>;
+ assigned-clock-parents = <&bpmp TEGRA186_CLK_PLL_A_OUT0>;
+ assigned-clock-rates = <12288000>;
+ sound-name-prefix = "DSPK1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra20-ac97.txt b/Bindings/sound/nvidia,tegra20-ac97.txt
new file mode 100644
index 0000000..eaf0010
--- /dev/null
+++ b/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/Bindings/sound/nvidia,tegra20-das.txt b/Bindings/sound/nvidia,tegra20-das.txt
new file mode 100644
index 0000000..6de3a7e
--- /dev/null
+++ b/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/Bindings/sound/nvidia,tegra20-i2s.yaml b/Bindings/sound/nvidia,tegra20-i2s.yaml
new file mode 100644
index 0000000..68ae124
--- /dev/null
+++ b/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#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra20 I2S Controller
+
+description: |
+ The I2S Controller streams synchronous serial audio data between system
+ memory and an external audio device. The controller supports the I2S Left
+ Justified Mode, Right Justified Mode, and DSP mode formats.
+
+maintainers:
+ - Thierry Reding <treding@nvidia.com>
+ - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+ compatible:
+ const: nvidia,tegra20-i2s
+
+ reg:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ reset-names:
+ const: i2s
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ minItems: 1
+
+ dmas:
+ minItems: 2
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+ nvidia,fixed-parent-rate:
+ description: |
+ Specifies whether board prefers parent clock to stay at a fixed rate.
+ This allows multiple Tegra20 audio components work simultaneously by
+ limiting number of supportable audio rates.
+ type: boolean
+
+required:
+ - compatible
+ - reg
+ - resets
+ - reset-names
+ - interrupts
+ - clocks
+ - dmas
+ - dma-names
+
+additionalProperties: false
+
+examples:
+ - |
+ i2s@70002800 {
+ compatible = "nvidia,tegra20-i2s";
+ reg = <0x70002800 0x200>;
+ interrupts = <45>;
+ clocks = <&tegra_car 11>;
+ resets = <&tegra_car 11>;
+ reset-names = "i2s";
+ dmas = <&apbdma 21>, <&apbdma 21>;
+ dma-names = "rx", "tx";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra20-spdif.yaml b/Bindings/sound/nvidia,tegra20-spdif.yaml
new file mode 100644
index 0000000..dc76a4d
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra20-spdif.yaml
@@ -0,0 +1,88 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra20-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra20 S/PDIF Controller
+
+description: |
+ The S/PDIF controller supports both input and output in serial audio
+ digital interface format. The input controller can digitally recover
+ a clock from the received stream. The S/PDIF controller is also used
+ to generate the embedded audio for HDMI output channel.
+
+maintainers:
+ - Thierry Reding <treding@nvidia.com>
+ - Jon Hunter <jonathanh@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: nvidia,tegra20-spdif
+
+ reg:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ minItems: 2
+
+ clock-names:
+ items:
+ - const: out
+ - const: in
+
+ dmas:
+ minItems: 2
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+ "#sound-dai-cells":
+ const: 0
+
+ nvidia,fixed-parent-rate:
+ description: |
+ Specifies whether board prefers parent clock to stay at a fixed rate.
+ This allows multiple Tegra20 audio components work simultaneously by
+ limiting number of supportable audio rates.
+ type: boolean
+
+required:
+ - compatible
+ - reg
+ - resets
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ spdif@70002400 {
+ compatible = "nvidia,tegra20-spdif";
+ reg = <0x70002400 0x200>;
+ interrupts = <77>;
+ clocks = <&clk 99>, <&clk 98>;
+ clock-names = "out", "in";
+ resets = <&rst 10>;
+ dmas = <&apbdma 3>, <&apbdma 3>;
+ dma-names = "rx", "tx";
+ #sound-dai-cells = <0>;
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-admaif.yaml b/Bindings/sound/nvidia,tegra210-admaif.yaml
new file mode 100644
index 0000000..15ab40a
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-admaif.yaml
@@ -0,0 +1,129 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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)$"
+ description:
+ Should be "rx1", "rx2" ... "rx20" for DMA Rx channel
+ Should be "tx1", "tx2" ... "tx20" for DMA Tx channel
+ minItems: 1
+ maxItems: 40
+
+required:
+ - compatible
+ - reg
+ - dmas
+ - dma-names
+
+additionalProperties: false
+
+examples:
+ - |
+ admaif@702d0000 {
+ compatible = "nvidia,tegra210-admaif";
+ reg = <0x702d0000 0x800>;
+ dmas = <&adma 1>, <&adma 1>,
+ <&adma 2>, <&adma 2>,
+ <&adma 3>, <&adma 3>,
+ <&adma 4>, <&adma 4>,
+ <&adma 5>, <&adma 5>,
+ <&adma 6>, <&adma 6>,
+ <&adma 7>, <&adma 7>,
+ <&adma 8>, <&adma 8>,
+ <&adma 9>, <&adma 9>,
+ <&adma 10>, <&adma 10>;
+ dma-names = "rx1", "tx1",
+ "rx2", "tx2",
+ "rx3", "tx3",
+ "rx4", "tx4",
+ "rx5", "tx5",
+ "rx6", "tx6",
+ "rx7", "tx7",
+ "rx8", "tx8",
+ "rx9", "tx9",
+ "rx10", "tx10";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-adx.yaml b/Bindings/sound/nvidia,tegra210-adx.yaml
new file mode 100644
index 0000000..e4c8717
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-adx.yaml
@@ -0,0 +1,77 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-adx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 ADX
+
+description: |
+ The Audio Demultiplexer (ADX) block takes an input stream with up to
+ 16 channels and demultiplexes it into four output streams of up to 16
+ channels each. A byte RAM helps to form output frames by any combination
+ of bytes from the input frame. Its design is identical to that of byte
+ RAM in the AMX except that the data flow direction is reversed.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^adx@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-adx
+ - items:
+ - enum:
+ - nvidia,tegra234-adx
+ - nvidia,tegra194-adx
+ - nvidia,tegra186-adx
+ - const: nvidia,tegra210-adx
+
+ reg:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^ADX[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ description: |
+ ADX has one input and four outputs. Accordingly ACIF (Audio Client
+ Interface) port nodes are defined to represent ADX input (port 0)
+ and outputs (ports 1 to 4). These are connected to corresponding
+ ports on AHUB (Audio Hub).
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: ADX ACIF input port
+ patternProperties:
+ '^port@[1-4]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: ADX ACIF output ports
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+
+ adx@702d3800 {
+ compatible = "nvidia,tegra210-adx";
+ reg = <0x702d3800 0x100>;
+ sound-name-prefix = "ADX1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-ahub.yaml b/Bindings/sound/nvidia,tegra210-ahub.yaml
new file mode 100644
index 0000000..c4abac8
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-ahub.yaml
@@ -0,0 +1,196 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ahub.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 AHUB
+
+description: |
+ The Audio Hub (AHUB) comprises a collection of hardware accelerators
+ for audio pre-processing, post-processing and a programmable full
+ crossbar for routing audio data across these accelerators. It has
+ external interfaces such as I2S, DMIC, DSPK. It interfaces with ADMA
+ engine through ADMAIF.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+ $nodename:
+ pattern: "^ahub@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - enum:
+ - nvidia,tegra210-ahub
+ - nvidia,tegra186-ahub
+ - nvidia,tegra234-ahub
+ - items:
+ - const: nvidia,tegra194-ahub
+ - const: nvidia,tegra186-ahub
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: ahub
+
+ assigned-clocks:
+ maxItems: 1
+
+ assigned-clock-parents:
+ maxItems: 1
+
+ assigned-clock-rates:
+ maxItems: 1
+
+ "#address-cells":
+ enum: [ 1, 2 ]
+
+ "#size-cells":
+ enum: [ 1, 2 ]
+
+ ranges: true
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ description: |
+ Contains list of ACIF (Audio CIF) port nodes for AHUB (Audio Hub).
+ These are connected to ACIF interfaces of AHUB clients. Thus the
+ number of port nodes depend on the number of clients that AHUB may
+ have depending on the SoC revision.
+
+ patternProperties:
+ '^port@[0-9]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+patternProperties:
+ '^i2s@[0-9a-f]+$':
+ type: object
+
+ '^dmic@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-dmic.yaml#
+
+ '^admaif@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-admaif.yaml#
+
+ '^dspk@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra186-dspk.yaml#
+
+ '^mvc@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-mvc.yaml#
+
+ '^sfc@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-sfc.yaml#
+
+ '^amx@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-amx.yaml#
+
+ '^adx@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-adx.yaml#
+
+ '^amixer@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-mixer.yaml#
+
+ '^asrc@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra186-asrc.yaml#
+
+ '^processing-engine@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-ope.yaml#
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - assigned-clocks
+ - assigned-clock-parents
+ - "#address-cells"
+ - "#size-cells"
+ - ranges
+
+additionalProperties: false
+
+examples:
+ - |
+ #include<dt-bindings/clock/tegra210-car.h>
+
+ ahub@702d0800 {
+ compatible = "nvidia,tegra210-ahub";
+ reg = <0x702d0800 0x800>;
+ clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
+ clock-names = "ahub";
+ assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
+ assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+ #address-cells = <1>;
+ #size-cells = <1>;
+ ranges = <0x702d0000 0x702d0000 0x0000e400>;
+
+ // All AHUB child nodes below
+ admaif@702d0000 {
+ compatible = "nvidia,tegra210-admaif";
+ reg = <0x702d0000 0x800>;
+ dmas = <&adma 1>, <&adma 1>,
+ <&adma 2>, <&adma 2>,
+ <&adma 3>, <&adma 3>,
+ <&adma 4>, <&adma 4>,
+ <&adma 5>, <&adma 5>,
+ <&adma 6>, <&adma 6>,
+ <&adma 7>, <&adma 7>,
+ <&adma 8>, <&adma 8>,
+ <&adma 9>, <&adma 9>,
+ <&adma 10>, <&adma 10>;
+ dma-names = "rx1", "tx1",
+ "rx2", "tx2",
+ "rx3", "tx3",
+ "rx4", "tx4",
+ "rx5", "tx5",
+ "rx6", "tx6",
+ "rx7", "tx7",
+ "rx8", "tx8",
+ "rx9", "tx9",
+ "rx10", "tx10";
+ };
+
+ i2s@702d1000 {
+ compatible = "nvidia,tegra210-i2s";
+ reg = <0x702d1000 0x100>;
+ clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+ clock-names = "i2s";
+ assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+ assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+ assigned-clock-rates = <1536000>;
+ sound-name-prefix = "I2S1";
+ };
+
+ dmic@702d4000 {
+ compatible = "nvidia,tegra210-dmic";
+ reg = <0x702d4000 0x100>;
+ clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+ clock-names = "dmic";
+ assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+ assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+ assigned-clock-rates = <3072000>;
+ sound-name-prefix = "DMIC1";
+ };
+
+ // More child nodes to follow
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-amx.yaml b/Bindings/sound/nvidia,tegra210-amx.yaml
new file mode 100644
index 0000000..021b725
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-amx.yaml
@@ -0,0 +1,79 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-amx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 AMX
+
+description: |
+ The Audio Multiplexer (AMX) block can multiplex up to four input streams
+ each of which can have maximum 16 channels and generate an output stream
+ with maximum 16 channels. A byte RAM helps to form an output frame by
+ any combination of bytes from the input frames.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^amx@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-amx
+ - items:
+ - const: nvidia,tegra186-amx
+ - const: nvidia,tegra210-amx
+ - const: nvidia,tegra194-amx
+ - items:
+ - const: nvidia,tegra234-amx
+ - const: nvidia,tegra194-amx
+
+ reg:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^AMX[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ description: |
+ AMX has four inputs and one output. Accordingly ACIF (Audio Client
+ Interfaces) port nodes are defined to represent AMX inputs (port 0
+ to 3) and output (port 4). These are connected to corresponding
+ ports on AHUB (Audio Hub).
+
+ patternProperties:
+ '^port@[0-3]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: AMX ACIF input ports
+
+ properties:
+ port@4:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: AMX ACIF output port
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+
+ amx@702d3000 {
+ compatible = "nvidia,tegra210-amx";
+ reg = <0x702d3000 0x100>;
+ sound-name-prefix = "AMX1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-dmic.yaml b/Bindings/sound/nvidia,tegra210-dmic.yaml
new file mode 100644
index 0000000..bff551c
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-dmic.yaml
@@ -0,0 +1,99 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-dmic.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 DMIC Controller
+
+description: |
+ The Digital MIC (DMIC) Controller is used to interface with Pulse
+ Density Modulation (PDM) input devices. It converts PDM signals to
+ Pulse Coded Modulation (PCM) signals. DMIC can be viewed as a PDM
+ receiver.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^dmic@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-dmic
+ - items:
+ - enum:
+ - nvidia,tegra234-dmic
+ - nvidia,tegra194-dmic
+ - nvidia,tegra186-dmic
+ - const: nvidia,tegra210-dmic
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: dmic
+
+ assigned-clocks:
+ maxItems: 1
+
+ assigned-clock-parents:
+ maxItems: 1
+
+ assigned-clock-rates:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^DMIC[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ DMIC ACIF (Audio Client Interface) port connected to the
+ corresponding AHUB (Audio Hub) ACIF port.
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ DMIC DAP (Digital Audio Port) interface which can be connected
+ to external audio codec for capture.
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - assigned-clocks
+ - assigned-clock-parents
+
+additionalProperties: false
+
+examples:
+ - |
+ #include<dt-bindings/clock/tegra210-car.h>
+
+ dmic@702d4000 {
+ compatible = "nvidia,tegra210-dmic";
+ reg = <0x702d4000 0x100>;
+ clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+ clock-names = "dmic";
+ assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+ assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+ assigned-clock-rates = <3072000>;
+ sound-name-prefix = "DMIC1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-i2s.yaml b/Bindings/sound/nvidia,tegra210-i2s.yaml
new file mode 100644
index 0000000..a82f11f
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-i2s.yaml
@@ -0,0 +1,115 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 I2S Controller
+
+description: |
+ The Inter-IC Sound (I2S) controller implements full-duplex,
+ bi-directional and single direction point-to-point serial
+ interfaces. It can interface with I2S compatible devices.
+ I2S controller can operate both in master and slave mode.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^i2s@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-i2s
+ - items:
+ - enum:
+ - nvidia,tegra234-i2s
+ - nvidia,tegra194-i2s
+ - nvidia,tegra186-i2s
+ - const: nvidia,tegra210-i2s
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ minItems: 1
+ items:
+ - description: I2S bit clock
+ - description:
+ Sync input clock, which can act as clock source to other I/O
+ modules in AHUB. The Tegra I2S driver sets this clock rate as
+ per bit clock rate. I/O module which wants to use this clock
+ as source, can mention this clock as parent in the DT bindings.
+ This is an optional clock entry, since it is only required when
+ some other I/O wants to reference from a particular I2Sx
+ instance.
+
+ clock-names:
+ minItems: 1
+ items:
+ - const: i2s
+ - const: sync_input
+
+ assigned-clocks:
+ minItems: 1
+ maxItems: 2
+
+ assigned-clock-parents:
+ minItems: 1
+ maxItems: 2
+
+ assigned-clock-rates:
+ minItems: 1
+ maxItems: 2
+
+ sound-name-prefix:
+ pattern: "^I2S[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ I2S ACIF (Audio Client Interface) port connected to the
+ corresponding AHUB (Audio Hub) ACIF port.
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ I2S DAP (Digital Audio Port) interface which can be connected
+ to external audio codec for playback or capture.
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - assigned-clocks
+ - assigned-clock-parents
+
+additionalProperties: false
+
+examples:
+ - |
+ #include<dt-bindings/clock/tegra210-car.h>
+
+ i2s@702d1000 {
+ compatible = "nvidia,tegra210-i2s";
+ reg = <0x702d1000 0x100>;
+ clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+ clock-names = "i2s";
+ assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+ assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+ assigned-clock-rates = <1536000>;
+ sound-name-prefix = "I2S1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-mbdrc.yaml b/Bindings/sound/nvidia,tegra210-mbdrc.yaml
new file mode 100644
index 0000000..5b91986
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-mbdrc.yaml
@@ -0,0 +1,47 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mbdrc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 MBDRC
+
+description:
+ The Multi Band Dynamic Range Compressor (MBDRC) is part of Output
+ Processing Engine (OPE) which interfaces with Audio Hub (AHUB) via
+ Audio Client Interface (ACIF). MBDRC can be used as a traditional
+ single full band or a dual band or a multi band dynamic processor.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-mbdrc
+ - items:
+ - enum:
+ - nvidia,tegra234-mbdrc
+ - nvidia,tegra194-mbdrc
+ - nvidia,tegra186-mbdrc
+ - const: nvidia,tegra210-mbdrc
+
+ reg:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ dynamic-range-compressor@702d8200 {
+ compatible = "nvidia,tegra210-mbdrc";
+ reg = <0x702d8200 0x200>;
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-mixer.yaml b/Bindings/sound/nvidia,tegra210-mixer.yaml
new file mode 100644
index 0000000..049898f
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-mixer.yaml
@@ -0,0 +1,75 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mixer.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 Mixer
+
+description: |
+ The Mixer supports mixing of up to ten 7.1 audio input streams and
+ generate five outputs (each of which can be any combination of the
+ ten input streams).
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^amixer@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-amixer
+ - items:
+ - enum:
+ - nvidia,tegra234-amixer
+ - nvidia,tegra194-amixer
+ - nvidia,tegra186-amixer
+ - const: nvidia,tegra210-amixer
+
+ reg:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^MIXER[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ description: |
+ Mixer has ten inputs and five outputs. Accordingly ACIF (Audio
+ Client Interfaces) port nodes are defined to represent Mixer
+ inputs (port 0 to 9) and outputs (port 10 to 14). These are
+ connected to corresponding ports on AHUB (Audio Hub).
+
+ patternProperties:
+ '^port@[0-9]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: Mixer ACIF input ports
+ '^port@[10-14]':
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: Mixer ACIF output ports
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+
+ amixer@702dbb00 {
+ compatible = "nvidia,tegra210-amixer";
+ reg = <0x702dbb00 0x800>;
+ sound-name-prefix = "MIXER1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-mvc.yaml b/Bindings/sound/nvidia,tegra210-mvc.yaml
new file mode 100644
index 0000000..d0280d8
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-mvc.yaml
@@ -0,0 +1,77 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mvc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 MVC
+
+description: |
+ The Master Volume Control (MVC) provides gain or attenuation to a digital
+ signal path. It can be used in input or output signal path for per-stream
+ volume control or it can be used as master volume control. The MVC block
+ has one input and one output. The input digital stream can be mono or
+ multi-channel (up to 7.1 channels) stream. An independent mute control is
+ also included in the MVC block.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^mvc@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-mvc
+ - items:
+ - enum:
+ - nvidia,tegra234-mvc
+ - nvidia,tegra194-mvc
+ - nvidia,tegra186-mvc
+ - const: nvidia,tegra210-mvc
+
+ reg:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^MVC[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ MVC ACIF (Audio Client Interface) input port. This is connected
+ to corresponding ACIF output port on AHUB (Audio Hub).
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ MVC ACIF output port. This is connected to corresponding ACIF
+ input port on AHUB.
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+
+ mvc@702da000 {
+ compatible = "nvidia,tegra210-mvc";
+ reg = <0x702da000 0x200>;
+ sound-name-prefix = "MVC1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-ope.yaml b/Bindings/sound/nvidia,tegra210-ope.yaml
new file mode 100644
index 0000000..9017fb6
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-ope.yaml
@@ -0,0 +1,87 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ope.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 OPE
+
+description:
+ The Output Processing Engine (OPE) is one of the AHUB client. It has
+ PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor)
+ sub blocks for data processing.
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-ope
+ - items:
+ - enum:
+ - nvidia,tegra234-ope
+ - nvidia,tegra194-ope
+ - nvidia,tegra186-ope
+ - const: nvidia,tegra210-ope
+
+ reg:
+ maxItems: 1
+
+ "#address-cells":
+ enum: [ 1, 2 ]
+
+ "#size-cells":
+ enum: [ 1, 2 ]
+
+ ranges: true
+
+ sound-name-prefix:
+ pattern: "^OPE[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description:
+ OPE ACIF (Audio Client Interface) input port. This is connected
+ to corresponding ACIF output port on AHUB (Audio Hub).
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description:
+ OPE ACIF output port. This is connected to corresponding ACIF
+ input port on AHUB.
+
+patternProperties:
+ '^equalizer@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-peq.yaml#
+
+ '^dynamic-range-compressor@[0-9a-f]+$':
+ type: object
+ $ref: nvidia,tegra210-mbdrc.yaml#
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ processing-engine@702d8000 {
+ compatible = "nvidia,tegra210-ope";
+ reg = <0x702d8000 0x100>;
+ sound-name-prefix = "OPE1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-peq.yaml b/Bindings/sound/nvidia,tegra210-peq.yaml
new file mode 100644
index 0000000..1e373c4
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-peq.yaml
@@ -0,0 +1,48 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-peq.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 PEQ
+
+description:
+ The Parametric Equalizer (PEQ) is a cascade of biquad filters with
+ each filter tuned based on certain parameters. It can be used to
+ equalize the irregularities in the speaker frequency response.
+ PEQ sits inside Output Processing Engine (OPE) which interfaces
+ with Audio Hub (AHUB) via Audio Client Interface (ACIF).
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-peq
+ - items:
+ - enum:
+ - nvidia,tegra234-peq
+ - nvidia,tegra194-peq
+ - nvidia,tegra186-peq
+ - const: nvidia,tegra210-peq
+
+ reg:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ equalizer@702d8100 {
+ compatible = "nvidia,tegra210-peq";
+ reg = <0x702d8100 0x100>;
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra210-sfc.yaml b/Bindings/sound/nvidia,tegra210-sfc.yaml
new file mode 100644
index 0000000..185ca0b
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra210-sfc.yaml
@@ -0,0 +1,74 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-sfc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 SFC
+
+description: |
+ The Sampling Frequency Converter (SFC) converts the sampling frequency
+ of the input signal from one frequency to another. It supports sampling
+ frequency conversions of streams of up to two channels (stereo).
+
+maintainers:
+ - Jon Hunter <jonathanh@nvidia.com>
+ - Mohan Kumar <mkumard@nvidia.com>
+ - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^sfc@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra210-sfc
+ - items:
+ - enum:
+ - nvidia,tegra234-sfc
+ - nvidia,tegra194-sfc
+ - nvidia,tegra186-sfc
+ - const: nvidia,tegra210-sfc
+
+ reg:
+ maxItems: 1
+
+ sound-name-prefix:
+ pattern: "^SFC[1-9]$"
+
+ ports:
+ $ref: /schemas/graph.yaml#/properties/ports
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ SFC ACIF (Audio Client Interface) input port. This is connected
+ to corresponding ACIF output port on AHUB (Audio Hub).
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+ description: |
+ SFC ACIF output port. This is connected to corresponding ACIF
+ input port on AHUB.
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+
+ sfc@702d2000 {
+ compatible = "nvidia,tegra210-sfc";
+ reg = <0x702d2000 0x200>;
+ sound-name-prefix = "SFC1";
+ };
+
+...
diff --git a/Bindings/sound/nvidia,tegra30-ahub.txt b/Bindings/sound/nvidia,tegra30-ahub.txt
new file mode 100644
index 0000000..0e9a189
--- /dev/null
+++ b/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/Bindings/sound/nvidia,tegra30-hda.yaml b/Bindings/sound/nvidia,tegra30-hda.yaml
new file mode 100644
index 0000000..12c31b4
--- /dev/null
+++ b/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/Bindings/sound/nvidia,tegra30-i2s.txt b/Bindings/sound/nvidia,tegra30-i2s.txt
new file mode 100644
index 0000000..38caa93
--- /dev/null
+++ b/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/Bindings/sound/nxp,tfa9879.yaml b/Bindings/sound/nxp,tfa9879.yaml
new file mode 100644
index 0000000..df26248
--- /dev/null
+++ b/Bindings/sound/nxp,tfa9879.yaml
@@ -0,0 +1,44 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nxp,tfa9879.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP TFA9879 class-D audio amplifier
+
+maintainers:
+ - Peter Rosin <peda@axentia.se>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: nxp,tfa9879
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c1 {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ amplifier@6c {
+ compatible = "nxp,tfa9879";
+ reg = <0x6c>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_i2c1>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/nxp,tfa989x.yaml b/Bindings/sound/nxp,tfa989x.yaml
new file mode 100644
index 0000000..fd2415e
--- /dev/null
+++ b/Bindings/sound/nxp,tfa989x.yaml
@@ -0,0 +1,99 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nxp,tfa989x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP/Goodix TFA989X (TFA1) Audio Amplifiers
+
+maintainers:
+ - Stephan Gerhold <stephan@gerhold.net>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - nxp,tfa9890
+ - nxp,tfa9895
+ - nxp,tfa9897
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ rcv-gpios:
+ description: optional GPIO to be asserted when receiver mode is enabled.
+
+ sound-name-prefix: true
+
+ vddd-supply:
+ description: regulator phandle for the VDDD power supply.
+
+if:
+ not:
+ properties:
+ compatible:
+ const: nxp,tfa9897
+then:
+ properties:
+ rcv-gpios: false
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@34 {
+ compatible = "nxp,tfa9895";
+ reg = <0x34>;
+ sound-name-prefix = "Speaker Left";
+ #sound-dai-cells = <0>;
+ };
+ audio-codec@36 {
+ compatible = "nxp,tfa9895";
+ reg = <0x36>;
+ sound-name-prefix = "Speaker Right";
+ #sound-dai-cells = <0>;
+ };
+ };
+
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ speaker_codec_top: audio-codec@34 {
+ compatible = "nxp,tfa9897";
+ reg = <0x34>;
+ vddd-supply = <&pm8916_l6>;
+ rcv-gpios = <&msmgpio 50 GPIO_ACTIVE_HIGH>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&speaker_top_default>;
+ sound-name-prefix = "Speaker Top";
+ #sound-dai-cells = <0>;
+ };
+
+ speaker_codec_bottom: audio-codec@36 {
+ compatible = "nxp,tfa9897";
+ reg = <0x36>;
+ vddd-supply = <&pm8916_l6>;
+ rcv-gpios = <&msmgpio 111 GPIO_ACTIVE_HIGH>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&speaker_bottom_default>;
+ sound-name-prefix = "Speaker Bottom";
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/omap-abe-twl6040.txt b/Bindings/sound/omap-abe-twl6040.txt
new file mode 100644
index 0000000..462b04e
--- /dev/null
+++ b/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/Bindings/sound/omap-dmic.txt b/Bindings/sound/omap-dmic.txt
new file mode 100644
index 0000000..418e30e
--- /dev/null
+++ b/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/Bindings/sound/omap-mcbsp.txt b/Bindings/sound/omap-mcbsp.txt
new file mode 100644
index 0000000..ae8bf70
--- /dev/null
+++ b/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/Bindings/sound/omap-mcpdm.txt b/Bindings/sound/omap-mcpdm.txt
new file mode 100644
index 0000000..ff98a0c
--- /dev/null
+++ b/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/Bindings/sound/omap-twl4030.txt b/Bindings/sound/omap-twl4030.txt
new file mode 100644
index 0000000..f6a715e
--- /dev/null
+++ b/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/Bindings/sound/option,gtm601.yaml b/Bindings/sound/option,gtm601.yaml
new file mode 100644
index 0000000..ff813d9
--- /dev/null
+++ b/Bindings/sound/option,gtm601.yaml
@@ -0,0 +1,42 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/option,gtm601.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: GTM601 UMTS modem audio interface CODEC
+
+maintainers:
+ - kernel@puri.sm
+
+description: >
+ This device has no configuration interface. The sample rate and channels are
+ based on the compatible string
+
+properties:
+ compatible:
+ oneOf:
+ - description: Broadmobi BM818 (48Khz stereo)
+ items:
+ - const: broadmobi,bm818
+ - const: option,gtm601
+ - description: GTM601 (8kHz mono)
+ const: option,gtm601
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+
+allOf:
+ - $ref: dai-common.yaml#
+
+additionalProperties: false
+
+examples:
+ - |
+ codec {
+ compatible = "option,gtm601";
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/pcm1789.txt b/Bindings/sound/pcm1789.txt
new file mode 100644
index 0000000..3c74ed2
--- /dev/null
+++ b/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/Bindings/sound/pcm179x.txt b/Bindings/sound/pcm179x.txt
new file mode 100644
index 0000000..436c2b2
--- /dev/null
+++ b/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/Bindings/sound/pcm186x.txt b/Bindings/sound/pcm186x.txt
new file mode 100644
index 0000000..1087f48
--- /dev/null
+++ b/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/Bindings/sound/pcm3060.txt b/Bindings/sound/pcm3060.txt
new file mode 100644
index 0000000..97de669
--- /dev/null
+++ b/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/Bindings/sound/pcm5102a.txt b/Bindings/sound/pcm5102a.txt
new file mode 100644
index 0000000..c63ab0b
--- /dev/null
+++ b/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/Bindings/sound/pcm512x.txt b/Bindings/sound/pcm512x.txt
new file mode 100644
index 0000000..77006a4
--- /dev/null
+++ b/Bindings/sound/pcm512x.txt
@@ -0,0 +1,53 @@
+PCM512x and TAS575x audio CODECs/amplifiers
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board). The TAS575x devices only support I2C.
+
+Required properties:
+
+ - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141",
+ "ti,pcm5142", "ti,tas5754" or "ti,tas5756"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+ - 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.
+ Caution: the TAS-desvices only support gpios 1,2 and 3
+
+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/Bindings/sound/qcom,apq8096.txt b/Bindings/sound/qcom,apq8096.txt
new file mode 100644
index 0000000..e1b9fa8
--- /dev/null
+++ b/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/Bindings/sound/qcom,lpass-cpu.yaml b/Bindings/sound/qcom,lpass-cpu.yaml
new file mode 100644
index 0000000..3a559bd
--- /dev/null
+++ b/Bindings/sound/qcom,lpass-cpu.yaml
@@ -0,0 +1,290 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-cpu.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Technologies Inc. LPASS CPU dai driver
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+ - Rohit kumar <quic_rohkumar@quicinc.com>
+
+description: |
+ Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist
+ of MI2S interface for audio data transfer on external codecs. LPASS cpu driver
+ is a module to configure Low-Power Audio Interface(LPAIF) core registers
+ across different IP versions.
+
+properties:
+ compatible:
+ enum:
+ - qcom,lpass-cpu
+ - qcom,apq8016-lpass-cpu
+ - qcom,sc7180-lpass-cpu
+ - qcom,sc7280-lpass-cpu
+
+ reg:
+ minItems: 1
+ maxItems: 6
+ description: LPAIF core registers
+
+ reg-names:
+ minItems: 1
+ maxItems: 6
+
+ clocks:
+ minItems: 3
+ maxItems: 10
+
+ clock-names:
+ minItems: 1
+ maxItems: 10
+
+ interrupts:
+ minItems: 1
+ maxItems: 4
+ description: LPAIF DMA buffer interrupt
+
+ interrupt-names:
+ minItems: 1
+ maxItems: 4
+
+ qcom,adsp:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: Phandle for the audio DSP node
+
+ iommus:
+ minItems: 2
+ maxItems: 3
+ description: Phandle to apps_smmu node with sid mask
+
+ power-domains:
+ maxItems: 1
+
+ power-domain-names:
+ maxItems: 1
+
+ required-opps:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 1
+
+ '#address-cells':
+ const: 1
+
+ '#size-cells':
+ const: 0
+
+patternProperties:
+ "^dai-link@[0-9a-f]+$":
+ type: object
+ description: |
+ LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node
+ depends on the specific driver providing the functionality and
+ properties.
+ properties:
+ reg:
+ maxItems: 1
+ description: Must be one of the DAI ID
+
+ qcom,playback-sd-lines:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description: list of MI2S data lines for playback
+
+ qcom,capture-sd-lines:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description: list of MI2S data lines for capture
+
+ required:
+ - reg
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - reg
+ - reg-names
+ - clocks
+ - clock-names
+ - interrupts
+ - interrupt-names
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: qcom,lpass-cpu
+
+ then:
+ properties:
+ clocks:
+ maxItems: 3
+ clock-names:
+ items:
+ - const: ahbix-clk
+ - const: mi2s-osr-clk
+ - const: mi2s-bit-clk
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: qcom,apq8016-lpass-cpu
+
+ then:
+ properties:
+ clocks:
+ minItems: 7
+ maxItems: 7
+ clock-names:
+ items:
+ - const: ahbix-clk
+ - const: mi2s-bit-clk0
+ - const: mi2s-bit-clk1
+ - const: mi2s-bit-clk2
+ - const: mi2s-bit-clk3
+ - const: pcnoc-mport-clk
+ - const: pcnoc-sway-clk
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: qcom,sc7180-lpass-cpu
+
+ then:
+ properties:
+ clocks:
+ minItems: 6
+ maxItems: 6
+ clock-names:
+ items:
+ - const: pcnoc-sway-clk
+ - const: audio-core
+ - const: mclk0
+ - const: pcnoc-mport-clk
+ - const: mi2s-bit-clk0
+ - const: mi2s-bit-clk1
+ reg:
+ minItems: 2
+ maxItems: 2
+ reg-names:
+ items:
+ - const: lpass-hdmiif
+ - const: lpass-lpaif
+ interrupts:
+ minItems: 2
+ maxItems: 2
+ interrupt-names:
+ items:
+ - const: lpass-irq-lpaif
+ - const: lpass-irq-hdmi
+ required:
+ - iommus
+ - power-domains
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: qcom,sc7280-lpass-cpu
+
+ then:
+ properties:
+ clocks:
+ minItems: 10
+ maxItems: 10
+ clock-names:
+ items:
+ - const: aon_cc_audio_hm_h
+ - const: audio_cc_ext_mclk0
+ - const: core_cc_sysnoc_mport_core
+ - const: core_cc_ext_if0_ibit
+ - const: core_cc_ext_if1_ibit
+ - const: audio_cc_codec_mem
+ - const: audio_cc_codec_mem0
+ - const: audio_cc_codec_mem1
+ - const: audio_cc_codec_mem2
+ - const: aon_cc_va_mem0
+ reg:
+ minItems: 6
+ maxItems: 6
+ reg-names:
+ items:
+ - const: lpass-hdmiif
+ - const: lpass-lpaif
+ - const: lpass-rxtx-cdc-dma-lpm
+ - const: lpass-rxtx-lpaif
+ - const: lpass-va-lpaif
+ - const: lpass-va-cdc-dma-lpm
+ interrupts:
+ minItems: 4
+ maxItems: 4
+ interrupt-names:
+ items:
+ - const: lpass-irq-lpaif
+ - const: lpass-irq-hdmi
+ - const: lpass-irq-vaif
+ - const: lpass-irq-rxtxif
+ power-domain-names:
+ items:
+ - const: lcx
+
+ required:
+ - iommus
+ - power-domains
+
+examples:
+ - |
+ #include <dt-bindings/sound/sc7180-lpass.h>
+
+ soc {
+ #address-cells = <2>;
+ #size-cells = <2>;
+ lpass@62d80000 {
+ compatible = "qcom,sc7180-lpass-cpu";
+
+ reg = <0 0x62d87000 0 0x68000>,
+ <0 0x62f00000 0 0x29000>;
+ reg-names = "lpass-hdmiif",
+ "lpass-lpaif";
+ iommus = <&apps_smmu 0x1020 0>,
+ <&apps_smmu 0x1032 0>;
+ power-domains = <&lpass_hm 0>;
+
+ clocks = <&gcc 131>,
+ <&lpasscorecc 6>,
+ <&lpasscorecc 7>,
+ <&lpasscorecc 10>,
+ <&lpasscorecc 8>,
+ <&lpasscorecc 9>;
+
+ clock-names = "pcnoc-sway-clk", "audio-core",
+ "mclk0", "pcnoc-mport-clk",
+ "mi2s-bit-clk0", "mi2s-bit-clk1";
+
+ interrupts = <0 160 1>,
+ <0 268 1>;
+ interrupt-names = "lpass-irq-lpaif",
+ "lpass-irq-hdmi";
+ #sound-dai-cells = <1>;
+
+ #address-cells = <1>;
+ #size-cells = <0>;
+ /* Optional to set different MI2S SD lines */
+ dai-link@0 {
+ reg = <MI2S_PRIMARY>;
+ qcom,playback-sd-lines = <1>;
+ qcom,capture-sd-lines = <0>;
+ };
+ };
+ };
+
+...
diff --git a/Bindings/sound/qcom,lpass-rx-macro.yaml b/Bindings/sound/qcom,lpass-rx-macro.yaml
new file mode 100644
index 0000000..ec4b0ac
--- /dev/null
+++ b/Bindings/sound/qcom,lpass-rx-macro.yaml
@@ -0,0 +1,130 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-rx-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) RX Macro audio codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-rx-macro
+ - qcom,sm8250-lpass-rx-macro
+ - qcom,sm8450-lpass-rx-macro
+ - qcom,sm8550-lpass-rx-macro
+ - qcom,sc8280xp-lpass-rx-macro
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ clocks:
+ minItems: 3
+ maxItems: 5
+
+ clock-names:
+ minItems: 3
+ maxItems: 5
+
+ clock-output-names:
+ maxItems: 1
+
+ power-domains:
+ maxItems: 2
+
+ power-domain-names:
+ items:
+ - const: macro
+ - const: dcodec
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-rx-macro
+ then:
+ properties:
+ clock-names:
+ oneOf:
+ - items: # for ADSP based platforms
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+ - items: # for ADSP bypass based platforms
+ - const: mclk
+ - const: npl
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sc8280xp-lpass-rx-macro
+ - qcom,sm8250-lpass-rx-macro
+ - qcom,sm8450-lpass-rx-macro
+ then:
+ properties:
+ clocks:
+ minItems: 5
+ maxItems: 5
+ clock-names:
+ items:
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sm8550-lpass-rx-macro
+ then:
+ properties:
+ clocks:
+ minItems: 4
+ maxItems: 4
+ clock-names:
+ items:
+ - const: mclk
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ codec@3200000 {
+ compatible = "qcom,sm8250-lpass-rx-macro";
+ reg = <0x3200000 0x1000>;
+ #sound-dai-cells = <1>;
+ #clock-cells = <0>;
+ clocks = <&audiocc 0>,
+ <&audiocc 1>,
+ <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&vamacro>;
+ clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
+ clock-output-names = "mclk";
+ };
diff --git a/Bindings/sound/qcom,lpass-tx-macro.yaml b/Bindings/sound/qcom,lpass-tx-macro.yaml
new file mode 100644
index 0000000..962701e
--- /dev/null
+++ b/Bindings/sound/qcom,lpass-tx-macro.yaml
@@ -0,0 +1,153 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-tx-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) TX Macro audio codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-tx-macro
+ - qcom,sm6115-lpass-tx-macro
+ - qcom,sm8250-lpass-tx-macro
+ - qcom,sm8450-lpass-tx-macro
+ - qcom,sm8550-lpass-tx-macro
+ - qcom,sc8280xp-lpass-tx-macro
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ clocks:
+ minItems: 3
+ maxItems: 5
+
+ clock-names:
+ minItems: 3
+ maxItems: 5
+
+ clock-output-names:
+ maxItems: 1
+
+ power-domains:
+ maxItems: 2
+
+ power-domain-names:
+ items:
+ - const: macro
+ - const: dcodec
+
+ qcom,dmic-sample-rate:
+ description: dmic sample rate
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-tx-macro
+ then:
+ properties:
+ clock-names:
+ oneOf:
+ - items: # for ADSP based platforms
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+ - items: # for ADSP bypass based platforms
+ - const: mclk
+ - const: npl
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sc8280xp-lpass-tx-macro
+ - qcom,sm8250-lpass-tx-macro
+ - qcom,sm8450-lpass-tx-macro
+ then:
+ properties:
+ clocks:
+ minItems: 5
+ maxItems: 5
+ clock-names:
+ items:
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sm6115-lpass-tx-macro
+ then:
+ properties:
+ clocks:
+ minItems: 4
+ maxItems: 4
+ clock-names:
+ items:
+ - const: mclk
+ - const: npl
+ - const: dcodec
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sm8550-lpass-tx-macro
+ then:
+ properties:
+ clocks:
+ minItems: 4
+ maxItems: 4
+ clock-names:
+ items:
+ - const: mclk
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ codec@3220000 {
+ compatible = "qcom,sm8250-lpass-tx-macro";
+ reg = <0x3220000 0x1000>;
+ #sound-dai-cells = <1>;
+ #clock-cells = <0>;
+ clocks = <&aoncc 0>,
+ <&aoncc 1>,
+ <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&vamacro>;
+ clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
+ clock-output-names = "mclk";
+ qcom,dmic-sample-rate = <600000>;
+ };
diff --git a/Bindings/sound/qcom,lpass-va-macro.yaml b/Bindings/sound/qcom,lpass-va-macro.yaml
new file mode 100644
index 0000000..4a56108
--- /dev/null
+++ b/Bindings/sound/qcom,lpass-va-macro.yaml
@@ -0,0 +1,148 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) VA Macro audio codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-va-macro
+ - qcom,sm8250-lpass-va-macro
+ - qcom,sm8450-lpass-va-macro
+ - qcom,sm8550-lpass-va-macro
+ - qcom,sc8280xp-lpass-va-macro
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ clocks:
+ minItems: 1
+ maxItems: 4
+
+ clock-names:
+ minItems: 1
+ maxItems: 4
+
+ clock-output-names:
+ maxItems: 1
+
+ power-domains:
+ maxItems: 2
+
+ power-domain-names:
+ items:
+ - const: macro
+ - const: dcodec
+
+ qcom,dmic-sample-rate:
+ description: dmic sample rate
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ vdd-micb-supply:
+ description: phandle to voltage regulator of MIC Bias
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - clock-names
+ - clocks
+
+allOf:
+ - $ref: dai-common.yaml#
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: qcom,sc7280-lpass-va-macro
+ then:
+ properties:
+ clocks:
+ maxItems: 1
+ clock-names:
+ items:
+ - const: mclk
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: qcom,sm8250-lpass-va-macro
+ then:
+ properties:
+ clocks:
+ minItems: 3
+ maxItems: 3
+ clock-names:
+ items:
+ - const: mclk
+ - const: macro
+ - const: dcodec
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - qcom,sc8280xp-lpass-va-macro
+ - qcom,sm8450-lpass-va-macro
+ then:
+ properties:
+ clocks:
+ minItems: 4
+ maxItems: 4
+ clock-names:
+ items:
+ - const: mclk
+ - const: macro
+ - const: dcodec
+ - const: npl
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - qcom,sm8550-lpass-va-macro
+ then:
+ properties:
+ clocks:
+ minItems: 3
+ maxItems: 3
+ clock-names:
+ items:
+ - const: mclk
+ - const: macro
+ - const: dcodec
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ codec@3370000 {
+ compatible = "qcom,sm8250-lpass-va-macro";
+ reg = <0x3370000 0x1000>;
+ #sound-dai-cells = <1>;
+ #clock-cells = <0>;
+ clocks = <&aoncc 0>,
+ <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>;
+ clock-names = "mclk", "macro", "dcodec";
+ clock-output-names = "fsgen";
+ qcom,dmic-sample-rate = <600000>;
+ vdd-micb-supply = <&vreg_s4a_1p8>;
+ };
diff --git a/Bindings/sound/qcom,lpass-wsa-macro.yaml b/Bindings/sound/qcom,lpass-wsa-macro.yaml
new file mode 100644
index 0000000..eea7609
--- /dev/null
+++ b/Bindings/sound/qcom,lpass-wsa-macro.yaml
@@ -0,0 +1,130 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-wsa-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) VA Macro audio codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-wsa-macro
+ - qcom,sm8250-lpass-wsa-macro
+ - qcom,sm8450-lpass-wsa-macro
+ - qcom,sm8550-lpass-wsa-macro
+ - qcom,sc8280xp-lpass-wsa-macro
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ clocks:
+ minItems: 4
+ maxItems: 6
+
+ clock-names:
+ minItems: 4
+ maxItems: 6
+
+ clock-output-names:
+ maxItems: 1
+
+ qcom,dmic-sample-rate:
+ description: dmic sample rate
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ vdd-micb-supply:
+ description: phandle to voltage regulator of MIC Bias
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+allOf:
+ - $ref: dai-common.yaml#
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sc7280-lpass-wsa-macro
+ - qcom,sm8450-lpass-wsa-macro
+ - qcom,sc8280xp-lpass-wsa-macro
+ then:
+ properties:
+ clocks:
+ minItems: 5
+ maxItems: 5
+ clock-names:
+ items:
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sm8250-lpass-wsa-macro
+ then:
+ properties:
+ clocks:
+ minItems: 6
+ clock-names:
+ items:
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: va
+ - const: fsgen
+
+ - if:
+ properties:
+ compatible:
+ enum:
+ - qcom,sm8550-lpass-wsa-macro
+ then:
+ properties:
+ clocks:
+ minItems: 4
+ maxItems: 4
+ clock-names:
+ items:
+ - const: mclk
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/qcom,sm8250-lpass-aoncc.h>
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ codec@3240000 {
+ compatible = "qcom,sm8250-lpass-wsa-macro";
+ reg = <0x3240000 0x1000>;
+ #sound-dai-cells = <1>;
+ #clock-cells = <0>;
+ clocks = <&audiocc 1>,
+ <&audiocc 0>,
+ <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&aoncc LPASS_CDC_VA_MCLK>,
+ <&vamacro>;
+ clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen";
+ clock-output-names = "mclk";
+ };
diff --git a/Bindings/sound/qcom,msm8916-wcd-digital.txt b/Bindings/sound/qcom,msm8916-wcd-digital.txt
new file mode 100644
index 0000000..1c8e4cb
--- /dev/null
+++ b/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/Bindings/sound/qcom,pm8916-wcd-analog-codec.yaml b/Bindings/sound/qcom,pm8916-wcd-analog-codec.yaml
new file mode 100644
index 0000000..94e7a18
--- /dev/null
+++ b/Bindings/sound/qcom,pm8916-wcd-analog-codec.yaml
@@ -0,0 +1,153 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,pm8916-wcd-analog-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm PM8916 WCD Analog Audio Codec
+
+maintainers:
+ - Konrad Dybcio <konradybcio@kernel.org>
+
+description:
+ The analog WCD audio codec found on Qualcomm PM8916 PMIC.
+
+properties:
+ compatible:
+ const: qcom,pm8916-wcd-analog-codec
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 14
+
+ interrupt-names:
+ items:
+ - const: cdc_spk_cnp_int
+ - const: cdc_spk_clip_int
+ - const: cdc_spk_ocp_int
+ - const: mbhc_ins_rem_det1
+ - const: mbhc_but_rel_det
+ - const: mbhc_but_press_det
+ - const: mbhc_ins_rem_det
+ - const: mbhc_switch_int
+ - const: cdc_ear_ocp_int
+ - const: cdc_hphr_ocp_int
+ - const: cdc_hphl_ocp_det
+ - const: cdc_ear_cnp_int
+ - const: cdc_hphr_cnp_int
+ - const: cdc_hphl_cnp_int
+
+ vdd-cdc-io-supply:
+ description: 1.8V buck supply
+
+ vdd-cdc-tx-rx-cx-supply:
+ description: 1.8V SIDO buck supply
+
+ vdd-micbias-supply:
+ description: micbias supply
+
+ qcom,mbhc-vthreshold-low:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Array of 5 threshold voltages in mV for 5-button detection on
+ headset when MBHC is powered by an internal current source.
+ minItems: 5
+ maxItems: 5
+
+ qcom,mbhc-vthreshold-high:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Array of 5 threshold voltages in mV for 5-button detection on
+ headset when MBHC is powered from micbias.
+ minItems: 5
+ maxItems: 5
+
+ qcom,micbias-lvl:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Voltage (mV) for Mic Bias
+
+ qcom,hphl-jack-type-normally-open:
+ type: boolean
+ description:
+ True if the HPHL pin on the jack is NO (Normally Open), false if it's
+ NC (Normally Closed).
+
+ qcom,gnd-jack-type-normally-open:
+ type: boolean
+ description:
+ True if the GND pin on the jack is NO (Normally Open), false if it's
+ NC (Normally Closed).
+
+ qcom,micbias1-ext-cap:
+ type: boolean
+ description:
+ True if micbias1 has an external capacitor.
+
+ qcom,micbias2-ext-cap:
+ type: boolean
+ description:
+ True if micbias2 has an external capacitor.
+
+ "#sound-dai-cells":
+ const: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/spmi/spmi.h>
+
+ pmic@1 {
+ compatible = "qcom,pm8916", "qcom,spmi-pmic";
+ reg = <0x1 SPMI_USID>;
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@f000 {
+ compatible = "qcom,pm8916-wcd-analog-codec";
+ reg = <0xf000>;
+ qcom,mbhc-vthreshold-low = <75 150 237 450 500>;
+ qcom,mbhc-vthreshold-high = <75 150 237 450 500>;
+ interrupt-parent = <&spmi_bus>;
+ interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x1 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x2 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x3 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x4 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x5 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x6 IRQ_TYPE_NONE>,
+ <0x1 0xf0 0x7 IRQ_TYPE_NONE>,
+ <0x1 0xf1 0x0 IRQ_TYPE_NONE>,
+ <0x1 0xf1 0x1 IRQ_TYPE_NONE>,
+ <0x1 0xf1 0x2 IRQ_TYPE_NONE>,
+ <0x1 0xf1 0x3 IRQ_TYPE_NONE>,
+ <0x1 0xf1 0x4 IRQ_TYPE_NONE>,
+ <0x1 0xf1 0x5 IRQ_TYPE_NONE>;
+ interrupt-names = "cdc_spk_cnp_int",
+ "cdc_spk_clip_int",
+ "cdc_spk_ocp_int",
+ "mbhc_ins_rem_det1",
+ "mbhc_but_rel_det",
+ "mbhc_but_press_det",
+ "mbhc_ins_rem_det",
+ "mbhc_switch_int",
+ "cdc_ear_ocp_int",
+ "cdc_hphr_ocp_int",
+ "cdc_hphl_ocp_det",
+ "cdc_ear_cnp_int",
+ "cdc_hphr_cnp_int",
+ "cdc_hphl_cnp_int";
+ vdd-cdc-io-supply = <&pm8916_l5>;
+ vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>;
+ vdd-micbias-supply = <&pm8916_l13>;
+ #sound-dai-cells = <1>;
+ };
+ };
diff --git a/Bindings/sound/qcom,q6adm-routing.yaml b/Bindings/sound/qcom,q6adm-routing.yaml
new file mode 100644
index 0000000..3f11d2e
--- /dev/null
+++ b/Bindings/sound/qcom,q6adm-routing.yaml
@@ -0,0 +1,39 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6adm-routing.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Device Manager (Q6ADM) routing
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+ Qualcomm Audio Device Manager (Q6ADM) routing node represents routing
+ specific configuration.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6adm-routing
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ routing {
+ compatible = "qcom,q6adm-routing";
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/qcom,q6adm.yaml b/Bindings/sound/qcom,q6adm.yaml
new file mode 100644
index 0000000..fe14a97
--- /dev/null
+++ b/Bindings/sound/qcom,q6adm.yaml
@@ -0,0 +1,51 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6adm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Device Manager (Q6ADM)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: /schemas/soc/qcom/qcom,apr-services.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6adm
+
+ routing:
+ type: object
+ $ref: /schemas/sound/qcom,q6adm-routing.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP LPASS audio routing
+
+required:
+ - compatible
+ - routing
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/soc/qcom,apr.h>
+
+ apr {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ service@8 {
+ compatible = "qcom,q6adm";
+ reg = <APR_SVC_ADM>;
+ qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+ routing {
+ compatible = "qcom,q6adm-routing";
+ #sound-dai-cells = <0>;
+ };
+ };
+ };
diff --git a/Bindings/sound/qcom,q6afe.yaml b/Bindings/sound/qcom,q6afe.yaml
new file mode 100644
index 0000000..297aa36
--- /dev/null
+++ b/Bindings/sound/qcom,q6afe.yaml
@@ -0,0 +1,68 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6afe.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio FrontEnd (Q6AFE)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: /schemas/soc/qcom/qcom,apr-services.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6afe
+
+ clock-controller:
+ $ref: /schemas/sound/qcom,q6dsp-lpass-clocks.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP LPASS clock controller
+
+ dais:
+ type: object
+ $ref: /schemas/sound/qcom,q6dsp-lpass-ports.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP audio ports
+
+required:
+ - compatible
+ - dais
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/soc/qcom,apr.h>
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ apr {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ service@4 {
+ compatible = "qcom,q6afe";
+ reg = <APR_SVC_AFE>;
+ qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+ clock-controller {
+ compatible = "qcom,q6afe-clocks";
+ #clock-cells = <2>;
+ };
+
+ dais {
+ compatible = "qcom,q6afe-dais";
+ #address-cells = <1>;
+ #size-cells = <0>;
+ #sound-dai-cells = <1>;
+
+ dai@22 {
+ reg = <QUATERNARY_MI2S_RX>;
+ qcom,sd-lines = <0 1 2 3>;
+ };
+ };
+ };
+ };
diff --git a/Bindings/sound/qcom,q6apm-dai.yaml b/Bindings/sound/qcom,q6apm-dai.yaml
new file mode 100644
index 0000000..9e5b30d
--- /dev/null
+++ b/Bindings/sound/qcom,q6apm-dai.yaml
@@ -0,0 +1,34 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6apm-dai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Process Manager Digital Audio Interfaces
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ This binding describes the Qualcomm APM DAIs in DSP
+
+properties:
+ compatible:
+ const: qcom,q6apm-dais
+
+ iommus:
+ minItems: 1
+ maxItems: 2
+
+required:
+ - compatible
+ - iommus
+
+additionalProperties: false
+
+examples:
+ - |
+ dais {
+ compatible = "qcom,q6apm-dais";
+ iommus = <&apps_smmu 0x1801 0x0>;
+ };
diff --git a/Bindings/sound/qcom,q6apm-lpass-dais.yaml b/Bindings/sound/qcom,q6apm-lpass-dais.yaml
new file mode 100644
index 0000000..894e653
--- /dev/null
+++ b/Bindings/sound/qcom,q6apm-lpass-dais.yaml
@@ -0,0 +1,35 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6apm-lpass-dais.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm DSP LPASS (Low Power Audio SubSystem) Audio Ports
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6apm-lpass-dais
+
+ '#sound-dai-cells':
+ const: 1
+
+required:
+ - compatible
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ dais {
+ compatible = "qcom,q6apm-lpass-dais";
+ #sound-dai-cells = <1>;
+ };
diff --git a/Bindings/sound/qcom,q6apm.yaml b/Bindings/sound/qcom,q6apm.yaml
new file mode 100644
index 0000000..ef1965a
--- /dev/null
+++ b/Bindings/sound/qcom,q6apm.yaml
@@ -0,0 +1,68 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6apm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Process Manager (Q6APM)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+ - $ref: /schemas/soc/qcom/qcom,apr-services.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6apm
+
+ bedais:
+ type: object
+ $ref: /schemas/sound/qcom,q6apm-lpass-dais.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP audio ports
+
+ dais:
+ type: object
+ $ref: /schemas/sound/qcom,q6apm-dai.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP audio ports
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - bedais
+ - dais
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/soc/qcom,gpr.h>
+
+ gpr {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ service@1 {
+ reg = <GPR_APM_MODULE_IID>;
+ compatible = "qcom,q6apm";
+ #sound-dai-cells = <0>;
+ qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+ dais {
+ compatible = "qcom,q6apm-dais";
+ iommus = <&apps_smmu 0x1801 0x0>;
+ };
+
+ bedais {
+ compatible = "qcom,q6apm-lpass-dais";
+ #sound-dai-cells = <1>;
+ };
+ };
+ };
diff --git a/Bindings/sound/qcom,q6asm-dais.yaml b/Bindings/sound/qcom,q6asm-dais.yaml
new file mode 100644
index 0000000..ce81194
--- /dev/null
+++ b/Bindings/sound/qcom,q6asm-dais.yaml
@@ -0,0 +1,96 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6asm-dais.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Stream Manager (Q6ASM)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+ Q6ASM is one of the APR audio services on Q6DSP. Each of its subnodes
+ represent a dai with board specific configuration.
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6asm-dais
+
+ iommus:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 0
+
+patternProperties:
+ "^dai@[0-9]+$":
+ type: object
+ description:
+ Q6ASM Digital Audio Interface
+
+ properties:
+ reg:
+ maxItems: 1
+
+ direction:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1, 2]
+ description: |
+ The direction of the dai stream::
+ - Q6ASM_DAI_TX_RX (0) for both tx and rx
+ - Q6ASM_DAI_TX (1) for only tx (Capture/Encode)
+ - Q6ASM_DAI_RX (2) for only rx (Playback/Decode)
+
+ is-compress-dai:
+ type: boolean
+ description:
+ Compress offload dai.
+
+ dependencies:
+ is-compress-dai: [ direction ]
+
+ required:
+ - reg
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+ - "#address-cells"
+ - "#size-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ dais {
+ compatible = "qcom,q6asm-dais";
+ iommus = <&apps_smmu 0x1821 0x0>;
+ #address-cells = <1>;
+ #size-cells = <0>;
+ #sound-dai-cells = <1>;
+
+ dai@0 {
+ reg = <0>;
+ };
+
+ dai@1 {
+ reg = <1>;
+ };
+
+ dai@2 {
+ reg = <2>;
+ is-compress-dai;
+ direction = <1>;
+ };
+ };
diff --git a/Bindings/sound/qcom,q6asm.yaml b/Bindings/sound/qcom,q6asm.yaml
new file mode 100644
index 0000000..cb49f96
--- /dev/null
+++ b/Bindings/sound/qcom,q6asm.yaml
@@ -0,0 +1,68 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6asm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Stream Manager (Q6ASM)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: /schemas/soc/qcom/qcom,apr-services.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6asm
+
+ dais:
+ type: object
+ $ref: /schemas/sound/qcom,q6asm-dais.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP audio ports
+
+required:
+ - compatible
+ - dais
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/soc/qcom,apr.h>
+
+ apr {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ service@7 {
+ compatible = "qcom,q6asm";
+ reg = <APR_SVC_ASM>;
+ qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+ dais {
+ compatible = "qcom,q6asm-dais";
+ iommus = <&apps_smmu 0x1821 0x0>;
+ #address-cells = <1>;
+ #size-cells = <0>;
+ #sound-dai-cells = <1>;
+
+ dai@0 {
+ reg = <0>;
+ };
+
+ dai@1 {
+ reg = <1>;
+ };
+
+ dai@2 {
+ reg = <2>;
+ is-compress-dai;
+ direction = <1>;
+ };
+ };
+ };
+ };
diff --git a/Bindings/sound/qcom,q6core.yaml b/Bindings/sound/qcom,q6core.yaml
new file mode 100644
index 0000000..e240712
--- /dev/null
+++ b/Bindings/sound/qcom,q6core.yaml
@@ -0,0 +1,39 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6core.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Core (Q6Core)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: /schemas/soc/qcom/qcom,apr-services.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6core
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/soc/qcom,apr.h>
+
+ apr {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ service@3 {
+ compatible = "qcom,q6core";
+ reg = <APR_SVC_ADSP_CORE>;
+ qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+ };
+ };
diff --git a/Bindings/sound/qcom,q6dsp-lpass-clocks.yaml b/Bindings/sound/qcom,q6dsp-lpass-clocks.yaml
new file mode 100644
index 0000000..3552c44
--- /dev/null
+++ b/Bindings/sound/qcom,q6dsp-lpass-clocks.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-clocks.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm DSP LPASS Clock Controller
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ This binding describes the Qualcomm DSP Clock Controller
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6afe-clocks
+ - qcom,q6prm-lpass-clocks
+
+ '#clock-cells':
+ const: 2
+ description:
+ Clock Id is followed by clock coupling attributes.
+ 1 = for no coupled clock
+ 2 = for dividend of the coupled clock
+ 3 = for divisor of the coupled clock
+ 4 = for inverted and no couple clock
+
+required:
+ - compatible
+ - "#clock-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ clock-controller {
+ compatible = "qcom,q6afe-clocks";
+ #clock-cells = <2>;
+ };
diff --git a/Bindings/sound/qcom,q6dsp-lpass-ports.yaml b/Bindings/sound/qcom,q6dsp-lpass-ports.yaml
new file mode 100644
index 0000000..08c618e
--- /dev/null
+++ b/Bindings/sound/qcom,q6dsp-lpass-ports.yaml
@@ -0,0 +1,164 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-ports.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm DSP LPASS(Low Power Audio SubSystem) Audio Ports
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ This binding describes the Qualcomm DSP LPASS Audio ports
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6afe-dais
+
+ '#sound-dai-cells':
+ const: 1
+
+ '#address-cells':
+ const: 1
+
+ '#size-cells':
+ const: 0
+
+# Digital Audio Interfaces
+patternProperties:
+ '^dai@[0-9]+$':
+ type: object
+ description:
+ Q6DSP Digital Audio Interfaces.
+
+ properties:
+ reg:
+ description:
+ Digital Audio Interface ID
+
+ qcom,sd-lines:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ List of serial data lines used by this dai.should be one or more of the 0-3 sd lines.
+ minItems: 1
+ maxItems: 4
+ uniqueItems: true
+ items:
+ minimum: 0
+ maximum: 3
+
+ qcom,tdm-sync-mode:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1, 2]
+ description:
+ TDM Synchronization mode
+ 0 = Short sync bit mode
+ 1 = Long sync mode
+ 2 = Short sync slot mode
+
+ qcom,tdm-sync-src:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1]
+ description:
+ TDM Synchronization source
+ 0 = External source
+ 1 = Internal source
+
+ qcom,tdm-data-out:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1]
+ description:
+ TDM Data out signal to drive with other masters
+ 0 = Disable
+ 1 = Enable
+
+ qcom,tdm-invert-sync:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1]
+ description:
+ TDM Invert the sync
+ 0 = Normal
+ 1 = Invert
+
+ qcom,tdm-data-delay:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1, 2]
+ description:
+ TDM Number of bit clock to delay data
+ 0 = 0 bit clock cycle
+ 1 = 1 bit clock cycle
+ 2 = 2 bit clock cycle
+
+ qcom,tdm-data-align:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1]
+ description:
+ Indicate how data is packed within the slot. For example, 32 slot
+ width in case of sample bit width is 24TDM Invert the sync.
+ 0 = MSB
+ 1 = LSB
+
+ required:
+ - reg
+
+ allOf:
+ - if:
+ properties:
+ reg:
+ contains:
+ # TDM DAI ID range from PRIMARY_TDM_RX_0 - QUINARY_TDM_TX_7
+ items:
+ minimum: 24
+ maximum: 103
+ then:
+ required:
+ - qcom,tdm-sync-mode
+ - qcom,tdm-sync-src
+ - qcom,tdm-data-out
+ - qcom,tdm-invert-sync
+ - qcom,tdm-data-delay
+ - qcom,tdm-data-align
+
+ - if:
+ properties:
+ reg:
+ contains:
+ # MI2S DAI ID range PRIMARY_MI2S_RX - QUATERNARY_MI2S_TX and
+ # QUINARY_MI2S_RX - QUINARY_MI2S_TX
+ items:
+ oneOf:
+ - minimum: 16
+ maximum: 23
+ - minimum: 127
+ maximum: 128
+ then:
+ required:
+ - qcom,sd-lines
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+ - "#address-cells"
+ - "#size-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/qcom,q6dsp-lpass-ports.h>
+
+ dais {
+ compatible = "qcom,q6afe-dais";
+ #address-cells = <1>;
+ #size-cells = <0>;
+ #sound-dai-cells = <1>;
+
+ dai@22 {
+ reg = <QUATERNARY_MI2S_RX>;
+ qcom,sd-lines = <0 1 2 3>;
+ };
+ };
diff --git a/Bindings/sound/qcom,q6prm.yaml b/Bindings/sound/qcom,q6prm.yaml
new file mode 100644
index 0000000..f6dbb12
--- /dev/null
+++ b/Bindings/sound/qcom,q6prm.yaml
@@ -0,0 +1,50 @@
+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6prm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Proxy Resource Manager (Q6PRM)
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+allOf:
+ - $ref: /schemas/soc/qcom/qcom,apr-services.yaml#
+
+properties:
+ compatible:
+ enum:
+ - qcom,q6prm
+
+ clock-controller:
+ $ref: /schemas/sound/qcom,q6dsp-lpass-clocks.yaml#
+ unevaluatedProperties: false
+ description: Qualcomm DSP LPASS clock controller
+
+required:
+ - compatible
+ - clock-controller
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/soc/qcom,gpr.h>
+
+ gpr {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ service@2 {
+ reg = <GPR_PRM_MODULE_IID>;
+ compatible = "qcom,q6prm";
+ qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+ clock-controller {
+ compatible = "qcom,q6prm-lpass-clocks";
+ #clock-cells = <2>;
+ };
+ };
+ };
diff --git a/Bindings/sound/qcom,sm8250.yaml b/Bindings/sound/qcom,sm8250.yaml
new file mode 100644
index 0000000..e082a4f
--- /dev/null
+++ b/Bindings/sound/qcom,sm8250.yaml
@@ -0,0 +1,319 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,sm8250.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Technologies Inc. ASoC sound card drivers
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+ This bindings describes Qualcomm SoC based sound cards
+ which uses LPASS internal codec for audio.
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - lenovo,yoga-c630-sndcard
+ - qcom,db845c-sndcard
+ - const: qcom,sdm845-sndcard
+ - enum:
+ - qcom,apq8016-sbc-sndcard
+ - qcom,msm8916-qdsp6-sndcard
+ - qcom,qrb5165-rb5-sndcard
+ - qcom,sc7180-qdsp6-sndcard
+ - qcom,sc8280xp-sndcard
+ - qcom,sdm845-sndcard
+ - qcom,sm8250-sndcard
+ - qcom,sm8450-sndcard
+
+ 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/Bindings/sound/qcom,wcd9335.yaml b/Bindings/sound/qcom,wcd9335.yaml
new file mode 100644
index 0000000..34f8fe4
--- /dev/null
+++ b/Bindings/sound/qcom,wcd9335.yaml
@@ -0,0 +1,156 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wcd9335.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm WCD9335 Audio Codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+ Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC with in-built
+ Soundwire controller and interrupt mux. It supports both I2S/I2C and SLIMbus
+ audio interfaces.
+
+properties:
+ compatible:
+ const: slim217,1a0
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 2
+
+ clock-names:
+ items:
+ - const: mclk
+ - const: slimbus
+
+ interrupts:
+ maxItems: 2
+
+ interrupt-names:
+ items:
+ - const: intr1
+ - const: intr2
+
+ interrupt-controller: true
+
+ '#interrupt-cells':
+ const: 1
+
+ reset-gpios:
+ maxItems: 1
+
+ slim-ifc-dev:
+ description: SLIM IFC device interface
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ '#sound-dai-cells':
+ const: 1
+
+ vdd-buck-supply:
+ description: 1.8V buck supply
+
+ vdd-buck-sido-supply:
+ description: 1.8V SIDO buck supply
+
+ vdd-io-supply:
+ description: 1.8V I/O supply
+
+ vdd-micbias-supply:
+ description: micbias supply
+
+ vdd-rx-supply:
+ description: 1.8V rx supply
+
+ vdd-tx-supply:
+ description: 1.8V tx supply
+
+ vdd-vbat-supply:
+ description: vbat supply
+
+required:
+ - compatible
+ - reg
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ required:
+ - slim-ifc-dev
+ then:
+ required:
+ - clocks
+ - clock-names
+ - interrupts
+ - interrupt-names
+ - interrupt-controller
+ - '#interrupt-cells'
+ - reset-gpios
+ - slim-ifc-dev
+ - '#sound-dai-cells'
+ - vdd-buck-supply
+ - vdd-buck-sido-supply
+ - vdd-io-supply
+ - vdd-rx-supply
+ - vdd-tx-supply
+ else:
+ properties:
+ clocks: false
+ clock-names: false
+ interrupts: false
+ interrupt-names: false
+ interrupt-controller: false
+ '#interrupt-cells': false
+ reset-gpios: false
+ slim-ifc-dev: false
+ '#sound-dai-cells': false
+ vdd-buck-supply: false
+ vdd-buck-sido-supply: false
+ vdd-io-supply: false
+ vdd-micbias-supply: false
+ vdd-rx-supply: false
+ vdd-tx-supply: false
+ vdd-vbat-supply: false
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/qcom,rpmcc.h>
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ tasha_ifd: codec@0,0 {
+ compatible = "slim217,1a0";
+ reg = <0 0>;
+ };
+
+ codec@1,0 {
+ compatible = "slim217,1a0";
+ reg = <1 0>;
+
+ clock-names = "mclk", "slimbus";
+ clocks = <&div1_mclk>, <&rpmcc RPM_SMD_BB_CLK1>;
+
+ interrupt-parent = <&tlmm>;
+ interrupts = <54 IRQ_TYPE_LEVEL_HIGH>,
+ <53 IRQ_TYPE_LEVEL_HIGH>;
+ interrupt-names = "intr1", "intr2";
+ interrupt-controller;
+ #interrupt-cells = <1>;
+
+ reset-gpios = <&tlmm 64 GPIO_ACTIVE_LOW>;
+ slim-ifc-dev = <&tasha_ifd>;
+ #sound-dai-cells = <1>;
+
+ vdd-buck-supply = <&vreg_s4a_1p8>;
+ vdd-buck-sido-supply = <&vreg_s4a_1p8>;
+ vdd-tx-supply = <&vreg_s4a_1p8>;
+ vdd-rx-supply = <&vreg_s4a_1p8>;
+ vdd-io-supply = <&vreg_s4a_1p8>;
+ };
diff --git a/Bindings/sound/qcom,wcd934x.yaml b/Bindings/sound/qcom,wcd934x.yaml
new file mode 100644
index 0000000..4df59f3
--- /dev/null
+++ b/Bindings/sound/qcom,wcd934x.yaml
@@ -0,0 +1,238 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wcd934x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: 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:
+ description: IFC device interface
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ 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
+ additionalProperties: true
+ 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
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ required:
+ - slim-ifc-dev
+ then:
+ required:
+ - reset-gpios
+ - slim-ifc-dev
+ - interrupt-controller
+ - clock-frequency
+ - clock-output-names
+ - qcom,micbias1-microvolt
+ - qcom,micbias2-microvolt
+ - qcom,micbias3-microvolt
+ - qcom,micbias4-microvolt
+ - "#interrupt-cells"
+ - "#clock-cells"
+ - "#sound-dai-cells"
+ - "#address-cells"
+ - "#size-cells"
+ oneOf:
+ - required:
+ - interrupts-extended
+ - required:
+ - interrupts
+ else:
+ properties:
+ reset-gpios: false
+ slim-ifc-dev: false
+ interrupts: false
+ interrupt-controller: false
+ clock-frequency: false
+ clock-output-names: false
+ qcom,micbias1-microvolt: false
+ qcom,micbias2-microvolt: false
+ qcom,micbias3-microvolt: false
+ qcom,micbias4-microvolt: false
+ "#interrupt-cells": false
+ "#clock-cells": false
+ "#sound-dai-cells": false
+ "#address-cells": false
+ "#size-cells": false
+
+additionalProperties: false
+
+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/Bindings/sound/qcom,wcd938x-sdw.yaml b/Bindings/sound/qcom,wcd938x-sdw.yaml
new file mode 100644
index 0000000..b430dd3
--- /dev/null
+++ b/Bindings/sound/qcom,wcd938x-sdw.yaml
@@ -0,0 +1,70 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wcd938x-sdw.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm SoundWire Slave devices on WCD9380/WCD9385
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC.
+ It has RX and TX Soundwire slave devices. This bindings is for the
+ slave devices.
+
+properties:
+ compatible:
+ const: sdw20217010d00
+
+ reg:
+ maxItems: 1
+
+ qcom,tx-port-mapping:
+ description: |
+ Specifies static port mapping between slave and master tx ports.
+ In the order of slave port index.
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 4
+ maxItems: 4
+
+ qcom,rx-port-mapping:
+ description: |
+ Specifies static port mapping between slave and master rx ports.
+ In the order of slave port index.
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 5
+ maxItems: 5
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ soundwire@3210000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x03210000 0x2000>;
+ wcd938x_rx: codec@0,4 {
+ compatible = "sdw20217010d00";
+ reg = <0 4>;
+ qcom,rx-port-mapping = <1 2 3 4 5>;
+ };
+ };
+
+ soundwire@3230000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x03230000 0x2000>;
+ wcd938x_tx: codec@0,3 {
+ compatible = "sdw20217010d00";
+ reg = <0 3>;
+ qcom,tx-port-mapping = <2 3 4 5>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/qcom,wcd938x.yaml b/Bindings/sound/qcom,wcd938x.yaml
new file mode 100644
index 0000000..0185657
--- /dev/null
+++ b/Bindings/sound/qcom,wcd938x.yaml
@@ -0,0 +1,156 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wcd938x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm WCD9380/WCD9385 Audio Codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC.
+ It has RX and TX Soundwire slave devices.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ codec {
+ compatible = "qcom,wcd9380-codec";
+ reset-gpios = <&tlmm 32 0>;
+ #sound-dai-cells = <1>;
+ qcom,tx-device = <&wcd938x_tx>;
+ qcom,rx-device = <&wcd938x_rx>;
+ qcom,micbias1-microvolt = <1800000>;
+ qcom,micbias2-microvolt = <1800000>;
+ qcom,micbias3-microvolt = <1800000>;
+ qcom,micbias4-microvolt = <1800000>;
+ qcom,hphl-jack-type-normally-closed;
+ qcom,ground-jack-type-normally-closed;
+ qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>;
+ qcom,mbhc-headphone-vthreshold-microvolt = <50000>;
+ };
+
+ /* ... */
+
+ soundwire@3210000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x03210000 0x2000>;
+ wcd938x_rx: codec@0,4 {
+ compatible = "sdw20217010d00";
+ reg = <0 4>;
+ qcom,rx-port-mapping = <1 2 3 4 5>;
+ };
+ };
+
+ soundwire@3230000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x03230000 0x2000>;
+ wcd938x_tx: codec@0,3 {
+ compatible = "sdw20217010d00";
+ reg = <0 3>;
+ qcom,tx-port-mapping = <2 3 4 5>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/qcom,wsa881x.yaml b/Bindings/sound/qcom,wsa881x.yaml
new file mode 100644
index 0000000..ac03672
--- /dev/null
+++ b/Bindings/sound/qcom,wsa881x.yaml
@@ -0,0 +1,71 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wsa881x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: 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.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: sdw10217201000
+
+ reg:
+ maxItems: 1
+
+ powerdown-gpios:
+ description: GPIO spec for Powerdown/Shutdown line to use
+ maxItems: 1
+
+ '#thermal-sensor-cells':
+ const: 0
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - powerdown-gpios
+ - "#thermal-sensor-cells"
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ soundwire@c2d0000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x0c2d0000 0x2000>;
+
+ speaker@0,1 {
+ compatible = "sdw10217201000";
+ reg = <0 1>;
+ powerdown-gpios = <&wcdpinctrl 2 0>;
+ #thermal-sensor-cells = <0>;
+ #sound-dai-cells = <0>;
+ };
+
+ speaker@0,2 {
+ compatible = "sdw10217201000";
+ reg = <0 2>;
+ powerdown-gpios = <&wcdpinctrl 2 0>;
+ #thermal-sensor-cells = <0>;
+ #sound-dai-cells = <0>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/qcom,wsa883x.yaml b/Bindings/sound/qcom,wsa883x.yaml
new file mode 100644
index 0000000..ba572a7
--- /dev/null
+++ b/Bindings/sound/qcom,wsa883x.yaml
@@ -0,0 +1,81 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wsa883x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm WSA8830/WSA8832/WSA8835
+ smart speaker amplifier
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ WSA883X is the Qualcomm Aqstic smart speaker amplifier
+ Their primary operating mode uses a SoundWire digital audio
+ interface. This binding is for SoundWire interface.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: sdw10217020200
+
+ reg:
+ maxItems: 1
+
+ powerdown-gpios:
+ description: GPIO spec for Powerdown/Shutdown line to use (pin SD_N)
+ maxItems: 1
+
+ vdd-supply:
+ description: VDD Supply for the Codec
+
+ '#thermal-sensor-cells':
+ const: 0
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - vdd-supply
+ - powerdown-gpios
+ - "#thermal-sensor-cells"
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ soundwire-controller@3250000 {
+ #address-cells = <2>;
+ #size-cells = <0>;
+ reg = <0x3250000 0x2000>;
+
+ speaker@0,1 {
+ compatible = "sdw10217020200";
+ reg = <0 1>;
+ powerdown-gpios = <&tlmm 1 GPIO_ACTIVE_LOW>;
+ vdd-supply = <&vreg_s10b_1p8>;
+ #thermal-sensor-cells = <0>;
+ #sound-dai-cells = <0>;
+ sound-name-prefix = "SpkrLeft";
+ };
+
+ speaker@0,2 {
+ compatible = "sdw10217020200";
+ reg = <0 2>;
+ powerdown-gpios = <&tlmm 89 GPIO_ACTIVE_LOW>;
+ vdd-supply = <&vreg_s10b_1p8>;
+ #thermal-sensor-cells = <0>;
+ #sound-dai-cells = <0>;
+ sound-name-prefix = "SpkrRight";
+ };
+ };
+
+...
diff --git a/Bindings/sound/qcom,wsa8840.yaml b/Bindings/sound/qcom,wsa8840.yaml
new file mode 100644
index 0000000..e6723c9
--- /dev/null
+++ b/Bindings/sound/qcom,wsa8840.yaml
@@ -0,0 +1,66 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+ WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using
+ SoundWire digital audio interface.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: sdw20217020400
+
+ reg:
+ maxItems: 1
+
+ powerdown-gpios:
+ description: Powerdown/Shutdown line to use (pin SD_N)
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ vdd-1p8-supply: true
+ vdd-io-supply: true
+
+required:
+ - compatible
+ - reg
+ - powerdown-gpios
+ - '#sound-dai-cells'
+ - vdd-1p8-supply
+ - vdd-io-supply
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ soundwire-controller {
+ #address-cells = <2>;
+ #size-cells = <0>;
+
+ speaker@0,1 {
+ compatible = "sdw20217020400";
+ reg = <0 1>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&spkr_2_sd_n_active>;
+ powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>;
+ #sound-dai-cells = <0>;
+ sound-name-prefix = "SpkrRight";
+ vdd-1p8-supply = <&vreg_l15b_1p8>;
+ vdd-io-supply = <&vreg_l3g_1p2>;
+ };
+ };
diff --git a/Bindings/sound/realtek,alc5632.yaml b/Bindings/sound/realtek,alc5632.yaml
new file mode 100644
index 0000000..fb05988
--- /dev/null
+++ b/Bindings/sound/realtek,alc5632.yaml
@@ -0,0 +1,63 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/realtek,alc5632.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: ALC5632 audio CODEC
+
+description: |
+ Pins on the device (for linking into audio routes):
+ * SPK_OUTP
+ * SPK_OUTN
+ * HP_OUT_L
+ * HP_OUT_R
+ * AUX_OUT_P
+ * AUX_OUT_N
+ * LINE_IN_L
+ * LINE_IN_R
+ * PHONE_P
+ * PHONE_N
+ * MIC1_P
+ * MIC1_N
+ * MIC2_P
+ * MIC2_N
+ * MICBIAS1
+ * DMICDAT
+
+maintainers:
+ - Leon Romanovsky <leon@leon.nu>
+
+properties:
+ compatible:
+ const: realtek,alc5632
+
+ reg:
+ maxItems: 1
+
+ '#gpio-cells':
+ const: 2
+
+ gpio-controller: true
+
+required:
+ - compatible
+ - reg
+ - '#gpio-cells'
+ - gpio-controller
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "realtek,alc5632";
+ reg = <0x1a>;
+ gpio-controller;
+ #gpio-cells = <2>;
+ };
+ };
diff --git a/Bindings/sound/realtek,rt1015p.yaml b/Bindings/sound/realtek,rt1015p.yaml
new file mode 100644
index 0000000..7dac9e6
--- /dev/null
+++ b/Bindings/sound/realtek,rt1015p.yaml
@@ -0,0 +1,43 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/realtek,rt1015p.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Realtek rt1015p codec
+
+maintainers:
+ - Tzung-Bi Shih <tzungbi@kernel.org>
+
+description: |
+ Rt1015p is a rt1015 variant which does not support I2C and
+ only supports S24, 48kHz, 64FS.
+
+properties:
+ compatible:
+ enum:
+ - realtek,rt1015p
+ - realtek,rt1019p
+
+ sdb-gpios:
+ description:
+ GPIO used for shutdown control.
+ 0 means shut down; 1 means power on.
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ rt1015p: rt1015p {
+ compatible = "realtek,rt1015p";
+ sdb-gpios = <&pio 175 GPIO_ACTIVE_HIGH>;
+ };
diff --git a/Bindings/sound/realtek,rt1016.yaml b/Bindings/sound/realtek,rt1016.yaml
new file mode 100644
index 0000000..5287e9c
--- /dev/null
+++ b/Bindings/sound/realtek,rt1016.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/realtek,rt1016.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Reaktek RT1016 Stereo Class D Audio Amplifier
+
+maintainers:
+ - oder_chiou@realtek.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: realtek,rt1016
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "realtek,rt1016";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/realtek,rt5616.yaml b/Bindings/sound/realtek,rt5616.yaml
new file mode 100644
index 0000000..2483208
--- /dev/null
+++ b/Bindings/sound/realtek,rt5616.yaml
@@ -0,0 +1,49 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/realtek,rt5616.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Realtek rt5616 ALSA SoC audio codec driver
+
+description: |
+ Pins on the device (for linking into audio routes) for RT5616:
+
+ * IN1P
+ * IN2P
+ * IN2N
+ * LOUTL
+ * LOUTR
+ * HPOL
+ * HPOR
+
+maintainers:
+ - Bard Liao <bardliao@realtek.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: realtek,rt5616
+
+ reg:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@1b {
+ compatible = "realtek,rt5616";
+ reg = <0x1b>;
+ };
+ };
diff --git a/Bindings/sound/realtek,rt5682s.yaml b/Bindings/sound/realtek,rt5682s.yaml
new file mode 100644
index 0000000..ecfa7a5
--- /dev/null
+++ b/Bindings/sound/realtek,rt5682s.yaml
@@ -0,0 +1,150 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/realtek,rt5682s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Realtek rt5682s codec
+
+maintainers:
+ - Derek Fang <derek.fang@realtek.com>
+
+description: |
+ Rt5682s(ALC5682I-VS) is a rt5682i variant which supports I2C only.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: realtek,rt5682s
+
+ reg:
+ maxItems: 1
+ description: I2C address of the device.
+
+ interrupts:
+ maxItems: 1
+ description: The CODEC's interrupt output.
+
+ realtek,dmic1-data-pin:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # dmic1 data is not used
+ - 1 # using GPIO2 pin as dmic1 data pin
+ - 2 # using GPIO5 pin as dmic1 data pin
+ description: |
+ Specify which GPIO pin be used as DMIC1 data pin.
+
+ realtek,dmic1-clk-pin:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # dmic1 clk is not used
+ - 1 # using GPIO1 pin as dmic1 clock pin
+ - 2 # using GPIO3 pin as dmic1 clock pin
+ description: |
+ Specify which GPIO pin be used as DMIC1 clk pin.
+
+ realtek,jd-src:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # No JD is used
+ - 1 # using JD1 as JD source
+ description: |
+ Specify which JD source be used.
+
+ realtek,ldo1-en-gpios:
+ description: |
+ The GPIO that controls the CODEC's LDO1_EN pin.
+
+ realtek,dmic-clk-rate-hz:
+ description: |
+ Set the clock rate (hz) for the requirement of the particular DMIC.
+
+ realtek,dmic-delay-ms:
+ description: |
+ Set the delay time (ms) for the requirement of the particular DMIC.
+
+ realtek,amic-delay-ms:
+ description: |
+ Set the delay time (ms) for the requirement of the particular platform or AMIC.
+
+ realtek,dmic-clk-driving-high:
+ type: boolean
+ description: |
+ Set the high driving of the DMIC clock out.
+
+ clocks:
+ items:
+ - description: phandle and clock specifier for codec MCLK.
+
+ clock-names:
+ items:
+ - const: mclk
+
+ "#clock-cells":
+ const: 1
+
+ clock-output-names:
+ minItems: 2
+ maxItems: 2
+ description: Name given for DAI word clock and bit clock outputs.
+
+ "#sound-dai-cells":
+ const: 1
+
+ AVDD-supply:
+ description: Regulator supplying analog power through the AVDD pin.
+
+ MICVDD-supply:
+ description: Regulator supplying power for the microphone bias through the
+ MICVDD pin.
+
+ DBVDD-supply:
+ description: Regulator supplying I/O power through the DBVDD pin.
+
+ LDO1-IN-supply:
+ description: Regulator supplying power to the digital core and charge pump
+ through the LDO1_IN pin.
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - AVDD-supply
+ - MICVDD-supply
+ - DBVDD-supply
+ - LDO1-IN-supply
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "realtek,rt5682s";
+ reg = <0x1a>;
+ interrupts = <6 IRQ_TYPE_LEVEL_HIGH>;
+ realtek,ldo1-en-gpios =
+ <&gpio 2 GPIO_ACTIVE_HIGH>;
+ realtek,dmic1-data-pin = <1>;
+ realtek,dmic1-clk-pin = <1>;
+ realtek,jd-src = <1>;
+
+ #clock-cells = <1>;
+ clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk";
+
+ clocks = <&osc>;
+ clock-names = "mclk";
+
+ AVDD-supply = <&avdd_reg>;
+ MICVDD-supply = <&micvdd_reg>;
+ DBVDD-supply = <&dbvdd_reg>;
+ LDO1-IN-supply = <&ldo1_in_reg>;
+ };
+ };
diff --git a/Bindings/sound/renesas,fsi.yaml b/Bindings/sound/renesas,fsi.yaml
new file mode 100644
index 0000000..df91991
--- /dev/null
+++ b/Bindings/sound/renesas,fsi.yaml
@@ -0,0 +1,87 @@
+# 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>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ $nodename:
+ pattern: "^sound@.*"
+
+ compatible:
+ oneOf:
+ # for FSI2 SoC
+ - items:
+ - enum:
+ - renesas,fsi2-sh73a0 # SH-Mobile AG5
+ - renesas,fsi2-r8a7740 # R-Mobile A1
+ - enum:
+ - renesas,sh_fsi2
+ # for Generic
+ - items:
+ - enum:
+ - renesas,sh_fsi
+ - renesas,sh_fsi2
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ power-domains:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 1
+
+patternProperties:
+ "^fsi(a|b),spdif-connection$":
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: FSI is connected by S/PDIF
+
+ "^fsi(a|b),stream-mode-support$":
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: FSI supports 16bit stream mode
+
+ "^fsi(a|b),use-internal-clock$":
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: FSI uses internal clock when master mode
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - power-domains
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/r8a7740-clock.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ sh_fsi2: sound@fe1f0000 {
+ compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2";
+ reg = <0xfe1f0000 0x400>;
+ interrupts = <GIC_SPI 9 0x4>;
+ clocks = <&mstp3_clks R8A7740_CLK_FSI>;
+ power-domains = <&pd_a4mp>;
+
+ #sound-dai-cells = <1>;
+ fsia,spdif-connection;
+ fsia,stream-mode-support;
+ fsia,use-internal-clock;
+ };
diff --git a/Bindings/sound/renesas,idt821034.yaml b/Bindings/sound/renesas,idt821034.yaml
new file mode 100644
index 0000000..a2b92db
--- /dev/null
+++ b/Bindings/sound/renesas,idt821034.yaml
@@ -0,0 +1,75 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/renesas,idt821034.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Renesas IDT821034 codec device
+
+maintainers:
+ - Herve Codina <herve.codina@bootlin.com>
+
+description: |
+ The IDT821034 codec is a four channel PCM codec with onchip filters and
+ programmable gain setting.
+
+ The time-slots used by the codec must be set and so, the properties
+ 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and
+ 'dai-tdm-slot-rx-mask' must be present in the ALSA sound card node for
+ sub-nodes that involve the codec. The codec uses one 8bit time-slot per
+ channel.
+ 'dai-tdm-tdm-slot-with' must be set to 8.
+
+ The IDT821034 codec also supports 5 gpios (SLIC signals) per channel.
+
+allOf:
+ - $ref: /schemas/spi/spi-peripheral-props.yaml#
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: renesas,idt821034
+
+ reg:
+ description:
+ SPI device address.
+ maxItems: 1
+
+ spi-max-frequency:
+ maximum: 8192000
+
+ spi-cpha: true
+
+ '#sound-dai-cells':
+ const: 0
+
+ '#gpio-cells':
+ const: 2
+
+ gpio-controller: true
+
+required:
+ - compatible
+ - reg
+ - spi-cpha
+ - '#sound-dai-cells'
+ - gpio-controller
+ - '#gpio-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@0 {
+ compatible = "renesas,idt821034";
+ reg = <0>;
+ spi-max-frequency = <8192000>;
+ spi-cpha;
+ #sound-dai-cells = <0>;
+ gpio-controller;
+ #gpio-cells = <2>;
+ };
+ };
diff --git a/Bindings/sound/renesas,rsnd.txt b/Bindings/sound/renesas,rsnd.txt
new file mode 100644
index 0000000..dfd768b
--- /dev/null
+++ b/Bindings/sound/renesas,rsnd.txt
@@ -0,0 +1,255 @@
+Renesas R-Car sound
+
+=============================================
+* Modules
+=============================================
+
+Renesas R-Car and RZ/G sound is constructed from below modules
+(for Gen2 or later)
+
+ SCU : Sampling Rate Converter Unit
+ - SRC : Sampling Rate Converter
+ - CMD
+ - CTU : Channel Transfer Unit
+ - MIX : Mixer
+ - DVC : Digital Volume and Mute Function
+ SSIU : Serial Sound Interface Unit
+ SSI : Serial Sound Interface
+
+See detail of each module's channels, connection, limitation on datasheet
+
+=============================================
+* Multi channel
+=============================================
+
+Multi channel is supported by Multi-SSI, or TDM-SSI.
+
+ Multi-SSI : 6ch case, you can use stereo x 3 SSI
+ TDM-SSI : 6ch case, you can use TDM
+
+=============================================
+* Enable/Disable each modules
+=============================================
+
+See datasheet to check SRC/CTU/MIX/DVC connect-limitation.
+DT controls enabling/disabling module.
+${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 outputted 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/Bindings/sound/renesas,rsnd.yaml b/Bindings/sound/renesas,rsnd.yaml
new file mode 100644
index 0000000..13a5a0a
--- /dev/null
+++ b/Bindings/sound/renesas,rsnd.yaml
@@ -0,0 +1,542 @@
+# 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>
+
+definitions:
+ port-def:
+ $ref: audio-graph-port.yaml#/definitions/port-base
+ unevaluatedProperties: false
+ patternProperties:
+ "^endpoint(@[0-9a-f]+)?":
+ $ref: audio-graph-port.yaml#/definitions/endpoint-base
+ properties:
+ playback:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ capture:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ unevaluatedProperties: false
+
+properties:
+
+ compatible:
+ oneOf:
+ # for Gen1 SoC
+ - items:
+ - enum:
+ - renesas,rcar_sound-r8a7778 # R-Car M1A
+ - renesas,rcar_sound-r8a7779 # R-Car H1
+ - const: renesas,rcar_sound-gen1
+ # for Gen2 SoC
+ - items:
+ - enum:
+ - renesas,rcar_sound-r8a7742 # RZ/G1H
+ - renesas,rcar_sound-r8a7743 # RZ/G1M
+ - renesas,rcar_sound-r8a7744 # RZ/G1N
+ - renesas,rcar_sound-r8a7745 # RZ/G1E
+ - renesas,rcar_sound-r8a77470 # RZ/G1C
+ - renesas,rcar_sound-r8a7790 # R-Car H2
+ - renesas,rcar_sound-r8a7791 # R-Car M2-W
+ - renesas,rcar_sound-r8a7793 # R-Car M2-N
+ - renesas,rcar_sound-r8a7794 # R-Car E2
+ - const: renesas,rcar_sound-gen2
+ # for Gen3 SoC
+ - items:
+ - enum:
+ - renesas,rcar_sound-r8a774a1 # RZ/G2M
+ - renesas,rcar_sound-r8a774b1 # RZ/G2N
+ - renesas,rcar_sound-r8a774c0 # RZ/G2E
+ - renesas,rcar_sound-r8a774e1 # RZ/G2H
+ - renesas,rcar_sound-r8a7795 # R-Car H3
+ - renesas,rcar_sound-r8a7796 # R-Car M3-W
+ - renesas,rcar_sound-r8a77961 # R-Car M3-W+
+ - renesas,rcar_sound-r8a77965 # R-Car M3-N
+ - renesas,rcar_sound-r8a77990 # R-Car E3
+ - renesas,rcar_sound-r8a77995 # R-Car D3
+ - const: renesas,rcar_sound-gen3
+ # for Gen4 SoC
+ - items:
+ - const: renesas,rcar_sound-r8a779g0 # R-Car V4H
+ - const: renesas,rcar_sound-gen4
+ # for Generic
+ - enum:
+ - renesas,rcar_sound-gen1
+ - renesas,rcar_sound-gen2
+ - renesas,rcar_sound-gen3
+
+ reg:
+ minItems: 1
+ maxItems: 5
+
+ reg-names:
+ minItems: 1
+ maxItems: 5
+
+ "#sound-dai-cells":
+ description: |
+ it must be 0 if your system is using single DAI
+ it must be 1 if your system is using multi DAIs
+ This is used on simple-audio-card
+ enum: [0, 1]
+
+ "#clock-cells":
+ description: |
+ it must be 0 if your system has audio_clkout
+ it must be 1 if your system has audio_clkout0/1/2/3
+ enum: [0, 1]
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 0
+
+ clock-frequency:
+ description: for audio_clkout0/1/2/3
+
+ clkout-lr-asynchronous:
+ description: audio_clkoutn is asynchronizes with lr-clock.
+ $ref: /schemas/types.yaml#/definitions/flag
+
+ power-domains: true
+
+ resets:
+ minItems: 1
+ maxItems: 11
+
+ reset-names:
+ minItems: 1
+ maxItems: 11
+
+ clocks:
+ description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks.
+ minItems: 1
+ maxItems: 31
+
+ clock-names:
+ description: List of necessary clock names.
+ # details are defined below
+
+ # ports is below
+ port:
+ $ref: "#/definitions/port-def"
+
+ rcar_sound,dvc:
+ description: DVC subnode.
+ type: object
+ patternProperties:
+ "^dvc-[0-1]$":
+ type: object
+ additionalProperties: false
+
+ properties:
+ dmas:
+ maxItems: 1
+ dma-names:
+ const: tx
+ required:
+ - dmas
+ - dma-names
+ additionalProperties: false
+
+ rcar_sound,mix:
+ description: MIX subnode.
+ type: object
+ patternProperties:
+ "^mix-[0-1]$":
+ type: object
+ additionalProperties: false
+ additionalProperties: false
+
+ rcar_sound,ctu:
+ description: CTU subnode.
+ type: object
+ patternProperties:
+ "^ctu-[0-7]$":
+ type: object
+ additionalProperties: false
+ additionalProperties: false
+
+ rcar_sound,src:
+ description: SRC subnode.
+ type: object
+ patternProperties:
+ "^src-[0-9]$":
+ type: object
+ additionalProperties: false
+
+ properties:
+ interrupts:
+ maxItems: 1
+ dmas:
+ maxItems: 2
+ dma-names:
+ allOf:
+ - items:
+ enum:
+ - tx
+ - rx
+ additionalProperties: false
+
+ rcar_sound,ssiu:
+ description: SSIU subnode.
+ type: object
+ patternProperties:
+ "^ssiu-[0-9]+$":
+ type: object
+ additionalProperties: false
+
+ properties:
+ dmas:
+ maxItems: 2
+ dma-names:
+ allOf:
+ - items:
+ enum:
+ - tx
+ - rx
+ required:
+ - dmas
+ - dma-names
+ additionalProperties: false
+
+ rcar_sound,ssi:
+ description: SSI subnode.
+ type: object
+ patternProperties:
+ "^ssi-[0-9]$":
+ type: object
+ additionalProperties: false
+
+ properties:
+ interrupts:
+ maxItems: 1
+ dmas:
+ minItems: 2
+ maxItems: 4
+ dma-names:
+ allOf:
+ - items:
+ enum:
+ - tx
+ - rx
+ - txu # if no ssiu node
+ - rxu # if no ssiu node
+
+ shared-pin:
+ description: shared clock pin
+ $ref: /schemas/types.yaml#/definitions/flag
+ pio-transfer:
+ description: PIO transfer mode
+ $ref: /schemas/types.yaml#/definitions/flag
+ no-busif:
+ description: BUSIF is not used when [mem -> SSI] via DMA case
+ $ref: /schemas/types.yaml#/definitions/flag
+ required:
+ - interrupts
+ additionalProperties: false
+
+patternProperties:
+ # For DAI base
+ 'rcar_sound,dai(@[0-9a-f]+)?$':
+ description: DAI subnode.
+ type: object
+ patternProperties:
+ "^dai([0-9]+)?$":
+ type: object
+ additionalProperties: false
+
+ properties:
+ playback:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ capture:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ anyOf:
+ - required:
+ - playback
+ - required:
+ - capture
+ additionalProperties: false
+
+ 'ports(@[0-9a-f]+)?$':
+ $ref: audio-graph-port.yaml#/definitions/port-base
+ unevaluatedProperties: false
+ patternProperties:
+ '^port(@[0-9a-f]+)?$':
+ $ref: "#/definitions/port-def"
+
+required:
+ - compatible
+ - reg
+ - reg-names
+ - clocks
+ - clock-names
+
+allOf:
+ - $ref: dai-common.yaml#
+
+ # --------------------
+ # reg/reg-names
+ # --------------------
+ # for Gen1
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: renesas,rcar_sound-gen1
+ then:
+ properties:
+ reg:
+ maxItems: 3
+ reg-names:
+ items:
+ enum:
+ - scu
+ - ssi
+ - adg
+ # for Gen2/Gen3
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - renesas,rcar_sound-gen2
+ - renesas,rcar_sound-gen3
+ then:
+ properties:
+ reg:
+ minItems: 5
+ reg-names:
+ items:
+ enum:
+ - scu
+ - adg
+ - ssiu
+ - ssi
+ - audmapp
+ # for Gen4
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: renesas,rcar_sound-gen4
+ then:
+ properties:
+ reg:
+ maxItems: 4
+ reg-names:
+ items:
+ enum:
+ - adg
+ - ssiu
+ - ssi
+ - sdmc
+
+ # --------------------
+ # clock-names
+ # --------------------
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: renesas,rcar_sound-gen4
+ then:
+ properties:
+ clock-names:
+ maxItems: 3
+ items:
+ enum:
+ - ssi.0
+ - ssiu.0
+ - clkin
+ else:
+ properties:
+ clock-names:
+ minItems: 1
+ maxItems: 31
+ items:
+ oneOf:
+ - const: ssi-all
+ - pattern: '^ssi\.[0-9]$'
+ - pattern: '^src\.[0-9]$'
+ - pattern: '^mix\.[0-1]$'
+ - pattern: '^ctu\.[0-1]$'
+ - pattern: '^dvc\.[0-1]$'
+ - pattern: '^clk_(a|b|c|i)$'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/r8a7790-cpg-mssr.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/power/r8a7790-sysc.h>
+ rcar_sound: sound@ec500000 {
+ #sound-dai-cells = <1>;
+ compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2";
+ reg = <0xec500000 0x1000>, /* SCU */
+ <0xec5a0000 0x100>, /* ADG */
+ <0xec540000 0x1000>, /* SSIU */
+ <0xec541000 0x280>, /* SSI */
+ <0xec740000 0x200>; /* Audio DMAC peri peri*/
+ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp";
+
+ clocks = <&cpg CPG_MOD 1005>, /* SSI-ALL */
+ <&cpg CPG_MOD 1006>, <&cpg CPG_MOD 1007>, /* SSI9, SSI8 */
+ <&cpg CPG_MOD 1008>, <&cpg CPG_MOD 1009>, /* SSI7, SSI6 */
+ <&cpg CPG_MOD 1010>, <&cpg CPG_MOD 1011>, /* SSI5, SSI4 */
+ <&cpg CPG_MOD 1012>, <&cpg CPG_MOD 1013>, /* SSI3, SSI2 */
+ <&cpg CPG_MOD 1014>, <&cpg CPG_MOD 1015>, /* SSI1, SSI0 */
+ <&cpg CPG_MOD 1022>, <&cpg CPG_MOD 1023>, /* SRC9, SRC8 */
+ <&cpg CPG_MOD 1024>, <&cpg CPG_MOD 1025>, /* SRC7, SRC6 */
+ <&cpg CPG_MOD 1026>, <&cpg CPG_MOD 1027>, /* SRC5, SRC4 */
+ <&cpg CPG_MOD 1028>, <&cpg CPG_MOD 1029>, /* SRC3, SRC2 */
+ <&cpg CPG_MOD 1030>, <&cpg CPG_MOD 1031>, /* SRC1, SRC0 */
+ <&cpg CPG_MOD 1020>, <&cpg CPG_MOD 1021>, /* MIX1, MIX0 */
+ <&cpg CPG_MOD 1020>, <&cpg CPG_MOD 1021>, /* CTU1, CTU0 */
+ <&cpg CPG_MOD 1019>, <&cpg CPG_MOD 1018>, /* DVC0, DVC1 */
+ <&audio_clk_a>, <&audio_clk_b>, /* CLKA, CLKB */
+ <&audio_clk_c>, <&audio_clk_i>; /* CLKC, CLKI */
+
+ clock-names = "ssi-all",
+ "ssi.9", "ssi.8",
+ "ssi.7", "ssi.6",
+ "ssi.5", "ssi.4",
+ "ssi.3", "ssi.2",
+ "ssi.1", "ssi.0",
+ "src.9", "src.8",
+ "src.7", "src.6",
+ "src.5", "src.4",
+ "src.3", "src.2",
+ "src.1", "src.0",
+ "mix.1", "mix.0",
+ "ctu.1", "ctu.0",
+ "dvc.0", "dvc.1",
+ "clk_a", "clk_b",
+ "clk_c", "clk_i";
+
+ power-domains = <&sysc R8A7790_PD_ALWAYS_ON>;
+
+ resets = <&cpg 1005>,
+ <&cpg 1006>, <&cpg 1007>, <&cpg 1008>, <&cpg 1009>,
+ <&cpg 1010>, <&cpg 1011>, <&cpg 1012>, <&cpg 1013>,
+ <&cpg 1014>, <&cpg 1015>;
+ reset-names = "ssi-all",
+ "ssi.9", "ssi.8", "ssi.7", "ssi.6",
+ "ssi.5", "ssi.4", "ssi.3", "ssi.2",
+ "ssi.1", "ssi.0";
+
+ rcar_sound,dvc {
+ dvc0: dvc-0 {
+ dmas = <&audma0 0xbc>;
+ dma-names = "tx";
+ };
+ dvc1: dvc-1 {
+ dmas = <&audma0 0xbe>;
+ dma-names = "tx";
+ };
+ };
+
+ rcar_sound,mix {
+ mix0: mix-0 { };
+ mix1: mix-1 { };
+ };
+
+ rcar_sound,ctu {
+ ctu00: ctu-0 { };
+ ctu01: ctu-1 { };
+ ctu02: ctu-2 { };
+ ctu03: ctu-3 { };
+ ctu10: ctu-4 { };
+ ctu11: ctu-5 { };
+ ctu12: ctu-6 { };
+ ctu13: ctu-7 { };
+ };
+
+ rcar_sound,src {
+ src0: src-0 {
+ status = "disabled";
+ };
+ src1: src-1 {
+ interrupts = <GIC_SPI 353 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&audma0 0x87>, <&audma1 0x9c>;
+ dma-names = "rx", "tx";
+ };
+ /* skip after src-2 */
+ };
+
+ rcar_sound,ssiu {
+ ssiu00: ssiu-0 {
+ dmas = <&audma0 0x15>, <&audma1 0x16>;
+ dma-names = "rx", "tx";
+ };
+ ssiu01: ssiu-1 {
+ dmas = <&audma0 0x35>, <&audma1 0x36>;
+ dma-names = "rx", "tx";
+ };
+ /* skip after ssiu-2 */
+ };
+
+ rcar_sound,ssi {
+ ssi0: ssi-0 {
+ interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&audma0 0x01>, <&audma1 0x02>;
+ dma-names = "rx", "tx";
+ };
+ ssi1: ssi-1 {
+ interrupts = <GIC_SPI 371 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&audma0 0x03>, <&audma1 0x04>;
+ dma-names = "rx", "tx";
+ };
+ /* skip other ssi-2 */
+ };
+
+ /* DAI base */
+ rcar_sound,dai {
+ dai0 {
+ playback = <&ssi5>, <&src5>;
+ capture = <&ssi6>;
+ };
+ dai1 {
+ playback = <&ssi3>;
+ };
+ dai2 {
+ capture = <&ssi4>;
+ };
+ dai3 {
+ playback = <&ssi7>;
+ };
+ dai4 {
+ capture = <&ssi8>;
+ };
+ };
+
+ /* assume audio-graph */
+ port {
+ rsnd_endpoint: endpoint {
+ remote-endpoint = <&codec_endpoint>;
+
+ dai-format = "left_j";
+ bitclock-master = <&rsnd_endpoint0>;
+ frame-master = <&rsnd_endpoint0>;
+
+ playback = <&ssi0>, <&src0>, <&dvc0>;
+ capture = <&ssi1>, <&src1>, <&dvc1>;
+ };
+ };
+ };
+
+ /* assume audio-graph */
+ codec {
+ port {
+ codec_endpoint: endpoint {
+ remote-endpoint = <&rsnd_endpoint>;
+ };
+ };
+ };
diff --git a/Bindings/sound/renesas,rz-ssi.yaml b/Bindings/sound/renesas,rz-ssi.yaml
new file mode 100644
index 0000000..3b5ae45
--- /dev/null
+++ b/Bindings/sound/renesas,rz-ssi.yaml
@@ -0,0 +1,126 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/renesas,rz-ssi.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2)
+
+maintainers:
+ - Biju Das <biju.das.jz@bp.renesas.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ items:
+ - enum:
+ - renesas,r9a07g043-ssi # RZ/G2UL
+ - renesas,r9a07g044-ssi # RZ/G2{L,LC}
+ - renesas,r9a07g054-ssi # RZ/V2L
+ - const: renesas,rz-ssi
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ minItems: 2
+ maxItems: 3
+
+ interrupt-names:
+ oneOf:
+ - items:
+ - const: int_req
+ - const: dma_rx
+ - const: dma_tx
+ - items:
+ - const: int_req
+ - const: dma_rt
+
+ clocks:
+ maxItems: 4
+
+ clock-names:
+ items:
+ - const: ssi
+ - const: ssi_sfr
+ - const: audio_clk1
+ - const: audio_clk2
+
+ power-domains:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ dmas:
+ minItems: 1
+ maxItems: 2
+ description:
+ The first cell represents a phandle to dmac.
+ The second cell specifies the encoded MID/RID values of the SSI port
+ connected to the DMA client and the slave channel configuration
+ parameters.
+ bits[0:9] - Specifies MID/RID value of a SSI channel as below
+ MID/RID value of SSI rx0 = 0x256
+ MID/RID value of SSI tx0 = 0x255
+ MID/RID value of SSI rx1 = 0x25a
+ MID/RID value of SSI tx1 = 0x259
+ MID/RID value of SSI rt2 = 0x25f
+ MID/RID value of SSI rx3 = 0x262
+ MID/RID value of SSI tx3 = 0x261
+ bit[10] - HIEN = 1, Detects a request in response to the rising edge
+ of the signal
+ bit[11] - LVL = 0, Detects based on the edge
+ bits[12:14] - AM = 2, Bus cycle mode
+ bit[15] - TM = 0, Single transfer mode
+
+ dma-names:
+ oneOf:
+ - items:
+ - const: tx
+ - const: rx
+ - items:
+ - const: rt
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - interrupt-names
+ - clocks
+ - clock-names
+ - resets
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/r9a07g044-cpg.h>
+
+ ssi0: ssi@10049c00 {
+ compatible = "renesas,r9a07g044-ssi",
+ "renesas,rz-ssi";
+ reg = <0x10049c00 0x400>;
+ interrupts = <GIC_SPI 326 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 327 IRQ_TYPE_EDGE_RISING>,
+ <GIC_SPI 328 IRQ_TYPE_EDGE_RISING>;
+ interrupt-names = "int_req", "dma_rx", "dma_tx";
+ clocks = <&cpg CPG_MOD R9A07G044_SSI0_PCLK2>,
+ <&cpg CPG_MOD R9A07G044_SSI0_PCLK_SFR>,
+ <&audio_clk1>,
+ <&audio_clk2>;
+ clock-names = "ssi", "ssi_sfr", "audio_clk1", "audio_clk2";
+ power-domains = <&cpg>;
+ resets = <&cpg R9A07G044_SSI0_RST_M2_REG>;
+ dmas = <&dmac 0x2655>,
+ <&dmac 0x2656>;
+ dma-names = "tx", "rx";
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/richtek,rt9120.yaml b/Bindings/sound/richtek,rt9120.yaml
new file mode 100644
index 0000000..a1242e8
--- /dev/null
+++ b/Bindings/sound/richtek,rt9120.yaml
@@ -0,0 +1,62 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/richtek,rt9120.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Richtek RT9120 Class-D audio amplifier
+
+maintainers:
+ - ChiYuan Huang <cy_huang@richtek.com>
+
+description: |
+ The RT9120 is a high efficiency, I2S-input, stereo audio power amplifier
+ delivering 2*20W into 8 Ohm BTL speaker loads. It supports the wide input
+ voltage range from 4.5V to 26.4V to meet the need on most common
+ applications like as TV, monitors. home entertainment, electronic music
+ equipment.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - richtek,rt9120
+
+ reg:
+ description: I2C device address
+ maxItems: 1
+
+ pwdnn-gpios:
+ description: GPIO used for power down, low active
+ maxItems: 1
+
+ dvdd-supply:
+ description: |
+ Supply for the default on DVDD power, voltage domain must be 3P3V or 1P8V
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - dvdd-supply
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ rt9120@1a {
+ compatible = "richtek,rt9120";
+ reg = <0x1a>;
+ pwdnn-gpios = <&gpio26 2 0>;
+ dvdd-supply = <&vdd_io_reg>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/richtek,rtq9128.yaml b/Bindings/sound/richtek,rtq9128.yaml
new file mode 100644
index 0000000..d54686a
--- /dev/null
+++ b/Bindings/sound/richtek,rtq9128.yaml
@@ -0,0 +1,61 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/richtek,rtq9128.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Richtek RTQ9128 Automative Audio Power Amplifier
+
+maintainers:
+ - ChiYuan Huang <cy_huang@richtek.com>
+
+description:
+ The RTQ9128 is a ultra-low output noise, high-efficiency, four-channel
+ class-D audio power amplifier and delivering 4x75W into 4OHm at 10%
+ THD+N from a 25V supply in automotive applications.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - richtek,rtq9128
+
+ reg:
+ maxItems: 1
+
+ enable-gpios:
+ maxItems: 1
+
+ richtek,tdm-input-data2-select:
+ type: boolean
+ description:
+ By default, if TDM mode is used, TDM data input will select 'DATA1' pin
+ as the data source. This option will configure TDM data input source from
+ 'DATA1' to 'DATA2' pin.
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ speaker@1a {
+ compatible = "richtek,rtq9128";
+ reg = <0x1a>;
+ enable-gpios = <&gpio 26 GPIO_ACTIVE_HIGH>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/rockchip,i2s-tdm.yaml b/Bindings/sound/rockchip,i2s-tdm.yaml
new file mode 100644
index 0000000..7bb6c5d
--- /dev/null
+++ b/Bindings/sound/rockchip,i2s-tdm.yaml
@@ -0,0 +1,192 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip,i2s-tdm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip I2S/TDM Controller
+
+description:
+ The Rockchip I2S/TDM Controller is a Time Division Multiplexed
+ audio interface found in various Rockchip SoCs, allowing up
+ to 8 channels of audio over a serial interface.
+
+maintainers:
+ - Nicolas Frattaroli <frattaroli.nicolas@gmail.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - rockchip,px30-i2s-tdm
+ - rockchip,rk1808-i2s-tdm
+ - rockchip,rk3308-i2s-tdm
+ - rockchip,rk3568-i2s-tdm
+ - rockchip,rk3588-i2s-tdm
+ - rockchip,rv1126-i2s-tdm
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ dmas:
+ minItems: 1
+ maxItems: 2
+
+ dma-names:
+ minItems: 1
+ maxItems: 2
+ items:
+ enum:
+ - rx
+ - tx
+
+ clocks:
+ minItems: 3
+ items:
+ - description: clock for TX
+ - description: clock for RX
+ - description: AHB clock driving the interface
+ - description:
+ Parent clock for mclk_tx (only required when using mclk-calibrate)
+ - description:
+ Parent clock for mclk_rx (only required when using mclk-calibrate)
+ - description:
+ Clock for sample rates that are an integer multiple of 8000
+ (only required when using mclk-calibrate)
+ - description:
+ Clock for sample rates that are an integer multiple of 11025
+ (only required when using mclk-calibrate)
+
+ clock-names:
+ minItems: 3
+ items:
+ - const: mclk_tx
+ - const: mclk_rx
+ - const: hclk
+ - const: mclk_tx_src
+ - const: mclk_rx_src
+ - const: mclk_root0
+ - const: mclk_root1
+
+ resets:
+ minItems: 1
+ maxItems: 2
+ description: resets for the tx and rx directions
+
+ reset-names:
+ minItems: 1
+ maxItems: 2
+ items:
+ enum:
+ - tx-m
+ - rx-m
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+ power-domains:
+ maxItems: 1
+
+ rockchip,grf:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description:
+ The phandle of the syscon node for the GRF register.
+
+ rockchip,trcm-sync-tx-only:
+ type: boolean
+ description: Use TX BCLK/LRCK for both TX and RX.
+
+ rockchip,trcm-sync-rx-only:
+ type: boolean
+ description: Use RX BCLK/LRCK for both TX and RX.
+
+ "#sound-dai-cells":
+ const: 0
+
+ rockchip,i2s-rx-route:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Defines the mapping of I2S RX sdis to I2S data bus lines.
+ By default, they are mapped one-to-one.
+ rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0.
+ maxItems: 4
+ items:
+ enum: [0, 1, 2, 3]
+
+ rockchip,i2s-tx-route:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Defines the mapping of I2S TX sdos to I2S data bus lines.
+ By default, they are mapped one-to-one.
+ rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0.
+ maxItems: 4
+ items:
+ enum: [0, 1, 2, 3]
+
+ rockchip,io-multiplex:
+ description:
+ Specify that the GPIO lines on the I2S bus are multiplexed such that
+ the direction (input/output) needs to be dynamically adjusted.
+ type: boolean
+
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - dmas
+ - dma-names
+ - clocks
+ - clock-names
+ - resets
+ - reset-names
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/rk3568-cru.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/pinctrl/rockchip.h>
+
+ bus {
+ #address-cells = <2>;
+ #size-cells = <2>;
+ i2s@fe410000 {
+ compatible = "rockchip,rk3568-i2s-tdm";
+ reg = <0x0 0xfe410000 0x0 0x1000>;
+ interrupts = <GIC_SPI 53 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&cru MCLK_I2S1_8CH_TX>, <&cru MCLK_I2S1_8CH_RX>,
+ <&cru HCLK_I2S1_8CH>;
+ clock-names = "mclk_tx", "mclk_rx", "hclk";
+ dmas = <&dmac1 3>, <&dmac1 2>;
+ dma-names = "rx", "tx";
+ resets = <&cru SRST_M_I2S1_8CH_TX>, <&cru SRST_M_I2S1_8CH_RX>;
+ reset-names = "tx-m", "rx-m";
+ rockchip,trcm-sync-tx-only;
+ rockchip,grf = <&grf>;
+ #sound-dai-cells = <0>;
+ pinctrl-names = "default";
+ pinctrl-0 =
+ <&i2s1m0_sclktx
+ &i2s1m0_sclkrx
+ &i2s1m0_lrcktx
+ &i2s1m0_lrckrx
+ &i2s1m0_sdi0
+ &i2s1m0_sdi1
+ &i2s1m0_sdi2
+ &i2s1m0_sdi3
+ &i2s1m0_sdo0
+ &i2s1m0_sdo1
+ &i2s1m0_sdo2
+ &i2s1m0_sdo3>;
+ };
+ };
diff --git a/Bindings/sound/rockchip,pdm.yaml b/Bindings/sound/rockchip,pdm.yaml
new file mode 100644
index 0000000..ff9e400
--- /dev/null
+++ b/Bindings/sound/rockchip,pdm.yaml
@@ -0,0 +1,123 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip,pdm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip PDM controller
+
+description:
+ The Pulse Density Modulation Interface Controller (PDMC) is
+ a PDM interface controller and decoder that support PDM format.
+ It integrates a clock generator driving the PDM microphone
+ and embeds filters which decimate the incoming bit stream to
+ obtain most common audio rates.
+
+maintainers:
+ - Heiko Stuebner <heiko@sntech.de>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - rockchip,pdm
+ - rockchip,px30-pdm
+ - rockchip,rk1808-pdm
+ - rockchip,rk3308-pdm
+ - rockchip,rk3568-pdm
+ - rockchip,rv1126-pdm
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: clock for PDM controller
+ - description: clock for PDM BUS
+
+ clock-names:
+ items:
+ - const: pdm_clk
+ - const: pdm_hclk
+
+ dmas:
+ maxItems: 1
+
+ dma-names:
+ items:
+ - const: rx
+
+ power-domains:
+ maxItems: 1
+
+ resets:
+ items:
+ - description: reset for PDM controller
+
+ reset-names:
+ items:
+ - const: pdm-m
+
+ rockchip,path-map:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description:
+ Defines the mapping of PDM SDIx to PDM PATHx.
+ By default, they are mapped one-to-one.
+ maxItems: 4
+ uniqueItems: true
+ items:
+ enum: [ 0, 1, 2, 3 ]
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/rk3328-cru.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/pinctrl/rockchip.h>
+
+ bus {
+ #address-cells = <2>;
+ #size-cells = <2>;
+
+ pdm@ff040000 {
+ compatible = "rockchip,pdm";
+ reg = <0x0 0xff040000 0x0 0x1000>;
+ interrupts = <GIC_SPI 82 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&cru SCLK_PDM>, <&cru HCLK_PDM>;
+ clock-names = "pdm_clk", "pdm_hclk";
+ dmas = <&dmac 16>;
+ dma-names = "rx";
+ #sound-dai-cells = <0>;
+ pinctrl-names = "default", "sleep";
+ pinctrl-0 = <&pdmm0_clk
+ &pdmm0_sdi0
+ &pdmm0_sdi1
+ &pdmm0_sdi2
+ &pdmm0_sdi3>;
+ pinctrl-1 = <&pdmm0_clk_sleep
+ &pdmm0_sdi0_sleep
+ &pdmm0_sdi1_sleep
+ &pdmm0_sdi2_sleep
+ &pdmm0_sdi3_sleep>;
+ };
+ };
diff --git a/Bindings/sound/rockchip,rk3288-hdmi-analog.txt b/Bindings/sound/rockchip,rk3288-hdmi-analog.txt
new file mode 100644
index 0000000..73577ac
--- /dev/null
+++ b/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".
+
+Optional properties:
+- rockchip,hp-en-gpios = The phandle of the GPIO that power up/down the
+ headphone (when the analog output is an headphone).
+- rockchip,hp-det-gpios = The phandle of the GPIO that detects the headphone
+ (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/Bindings/sound/rockchip,rk3328-codec.yaml b/Bindings/sound/rockchip,rk3328-codec.yaml
new file mode 100644
index 0000000..5cdb8bc
--- /dev/null
+++ b/Bindings/sound/rockchip,rk3328-codec.yaml
@@ -0,0 +1,74 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip rk3328 internal codec
+
+maintainers:
+ - Heiko Stuebner <heiko@sntech.de>
+allOf:
+ - $ref: dai-common.yaml#
+
+
+properties:
+ compatible:
+ const: rockchip,rk3328-codec
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: clock for audio codec
+ - description: clock for I2S master clock
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: mclk
+
+ rockchip,grf:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description:
+ The phandle of the syscon node for the GRF register.
+
+ spk-depop-time-ms:
+ default: 200
+ description:
+ Speaker depop time in msec.
+
+ mute-gpios:
+ maxItems: 1
+ description:
+ GPIO specifier for external line driver control (typically the
+ dedicated GPIO_MUTE pin)
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - rockchip,grf
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/clock/rk3328-cru.h>
+ codec: codec@ff410000 {
+ compatible = "rockchip,rk3328-codec";
+ reg = <0xff410000 0x1000>;
+ clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>;
+ clock-names = "pclk", "mclk";
+ rockchip,grf = <&grf>;
+ mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
+ spk-depop-time-ms = <100>;
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/rockchip,rk3399-gru-sound.txt b/Bindings/sound/rockchip,rk3399-gru-sound.txt
new file mode 100644
index 0000000..72d3cf4
--- /dev/null
+++ b/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/Bindings/sound/rockchip-i2s.yaml b/Bindings/sound/rockchip-i2s.yaml
new file mode 100644
index 0000000..fcb01ab
--- /dev/null
+++ b/Bindings/sound/rockchip-i2s.yaml
@@ -0,0 +1,140 @@
+# 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>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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,rk3588-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
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+ 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"
+
+unevaluatedProperties: 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/Bindings/sound/rockchip-max98090.txt b/Bindings/sound/rockchip-max98090.txt
new file mode 100644
index 0000000..e9c58b2
--- /dev/null
+++ b/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/Bindings/sound/rockchip-rt5645.txt b/Bindings/sound/rockchip-rt5645.txt
new file mode 100644
index 0000000..411a62b
--- /dev/null
+++ b/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/Bindings/sound/rockchip-spdif.yaml b/Bindings/sound/rockchip-spdif.yaml
new file mode 100644
index 0000000..c3c989e
--- /dev/null
+++ b/Bindings/sound/rockchip-spdif.yaml
@@ -0,0 +1,105 @@
+# 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,rk3128-spdif
+ - 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"
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: rockchip,rk3288-spdif
+ then:
+ required:
+ - rockchip,grf
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/rk3188-cru.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ spdif: spdif@1011e000 {
+ compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
+ reg = <0x1011e000 0x2000>;
+ interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>;
+ clock-names = "mclk", "hclk";
+ dmas = <&dmac1_s 8>;
+ dma-names = "tx";
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/rohm,bd28623.yaml b/Bindings/sound/rohm,bd28623.yaml
new file mode 100644
index 0000000..5abcf92
--- /dev/null
+++ b/Bindings/sound/rohm,bd28623.yaml
@@ -0,0 +1,70 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: ROHM BD28623MUV Class D speaker amplifier for digital input
+
+description:
+ This codec does not have any control buses such as I2C, it detect
+ format and rate of I2S signal automatically. It has two signals
+ that can be connected to GPIOs reset and mute.
+
+maintainers:
+ - Katsuhiro Suzuki <katsuhiro@katsuster.net>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: rohm,bd28623
+
+ "#sound-dai-cells":
+ const: 0
+
+ VCCA-supply:
+ description:
+ regulator phandle for the VCCA (for analog) power supply
+
+ VCCP1-supply:
+ description:
+ regulator phandle for the VCCP1 (for ch1) power supply
+
+ VCCP2-supply:
+ description:
+ regulator phandle for the VCCP2 (for ch2) power supply
+
+ reset-gpios:
+ maxItems: 1
+ description:
+ GPIO specifier for the active low reset line
+
+ mute-gpios:
+ maxItems: 1
+ description:
+ GPIO specifier for the active low mute line
+
+required:
+ - compatible
+ - VCCA-supply
+ - VCCP1-supply
+ - VCCP2-supply
+ - "#sound-dai-cells"
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ codec {
+ compatible = "rohm,bd28623";
+ #sound-dai-cells = <0>;
+
+ VCCA-supply = <&vcc_reg>;
+ VCCP1-supply = <&vcc_reg>;
+ VCCP2-supply = <&vcc_reg>;
+ reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>;
+ mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>;
+ };
diff --git a/Bindings/sound/rt1011.txt b/Bindings/sound/rt1011.txt
new file mode 100644
index 0000000..02d53b9
--- /dev/null
+++ b/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/Bindings/sound/rt1015.txt b/Bindings/sound/rt1015.txt
new file mode 100644
index 0000000..e498966
--- /dev/null
+++ b/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/Bindings/sound/rt1019.yaml b/Bindings/sound/rt1019.yaml
new file mode 100644
index 0000000..3d5a91a
--- /dev/null
+++ b/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/Bindings/sound/rt1308.txt b/Bindings/sound/rt1308.txt
new file mode 100644
index 0000000..2d46084
--- /dev/null
+++ b/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/Bindings/sound/rt274.txt b/Bindings/sound/rt274.txt
new file mode 100644
index 0000000..791a1bd
--- /dev/null
+++ b/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/Bindings/sound/rt5514.txt b/Bindings/sound/rt5514.txt
new file mode 100644
index 0000000..d2cc171
--- /dev/null
+++ b/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/Bindings/sound/rt5631.txt b/Bindings/sound/rt5631.txt
new file mode 100644
index 0000000..56bc852
--- /dev/null
+++ b/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/Bindings/sound/rt5640.txt b/Bindings/sound/rt5640.txt
new file mode 100644
index 0000000..0c39858
--- /dev/null
+++ b/Bindings/sound/rt5640.txt
@@ -0,0 +1,97 @@
+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,lout-differential
+ Boolean. Indicate LOUT output is differential, rather than stereo.
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+- realtek,dmic1-data-pin
+ 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/Bindings/sound/rt5645.txt b/Bindings/sound/rt5645.txt
new file mode 100644
index 0000000..41a62fd
--- /dev/null
+++ b/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/Bindings/sound/rt5651.txt b/Bindings/sound/rt5651.txt
new file mode 100644
index 0000000..56e736a
--- /dev/null
+++ b/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/Bindings/sound/rt5659.txt b/Bindings/sound/rt5659.txt
new file mode 100644
index 0000000..8f3f62c
--- /dev/null
+++ b/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 dai-common.yaml
+
+- ports: A Codec may have a single or multiple I2S interfaces. These
+ interfaces on Codec side can be described under 'ports' or 'port'.
+ When the SoC or host device is connected to multiple interfaces of
+ the Codec, the connectivity can be described using 'ports' property.
+ If a single interface is used, then 'port' can be used. The usage
+ depends on the platform or board design.
+ Please refer to Documentation/devicetree/bindings/graph.txt
+
+Pins on the device (for linking into audio routes) for RT5659/RT5658:
+
+ * DMIC L1
+ * 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/Bindings/sound/rt5660.txt b/Bindings/sound/rt5660.txt
new file mode 100644
index 0000000..30be5f9
--- /dev/null
+++ b/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/Bindings/sound/rt5663.txt b/Bindings/sound/rt5663.txt
new file mode 100644
index 0000000..24a6dab
--- /dev/null
+++ b/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 maximum, volume, DC offset w/o and w/ mic of each L and
+ R channel accordingly. Example is shown as following.
+ < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0
+ 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0>
+ 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/Bindings/sound/rt5665.txt b/Bindings/sound/rt5665.txt
new file mode 100644
index 0000000..f6ca96b
--- /dev/null
+++ b/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/Bindings/sound/rt5668.txt b/Bindings/sound/rt5668.txt
new file mode 100644
index 0000000..a2b7e9a
--- /dev/null
+++ b/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/Bindings/sound/rt5677.txt b/Bindings/sound/rt5677.txt
new file mode 100644
index 0000000..da24300
--- /dev/null
+++ b/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/Bindings/sound/rt5682.txt b/Bindings/sound/rt5682.txt
new file mode 100644
index 0000000..5e1d08d
--- /dev/null
+++ b/Bindings/sound/rt5682.txt
@@ -0,0 +1,98 @@
+RT5682 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5682" or "realtek,rt5682i"
+
+- reg : The I2C address of the device.
+
+- AVDD-supply: phandle to the regulator supplying analog power through the
+ AVDD pin
+
+- MICVDD-supply: phandle to the regulator supplying power for the microphone
+ bias through the MICVDD pin. Either MICVDD or VBAT should be present.
+
+- VBAT-supply: phandle to the regulator supplying battery power through the
+ VBAT pin. Either MICVDD or VBAT should be present.
+
+- DBVDD-supply: phandle to the regulator supplying I/O power through the DBVDD
+ pin.
+
+- LDO1-IN-supply: phandle to the regulator supplying power to the digital core
+ and charge pump through the LDO1_IN pin.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+- 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";
+
+ AVDD-supply = <&avdd_reg>;
+ MICVDD-supply = <&micvdd_reg>;
+ DBVDD-supply = <&dbvdd_reg>;
+ LDO1-IN-supply = <&ldo1_in_reg>;
+};
diff --git a/Bindings/sound/samsung,aries-wm8994.yaml b/Bindings/sound/samsung,aries-wm8994.yaml
new file mode 100644
index 0000000..5ea0819
--- /dev/null
+++ b/Bindings/sound/samsung,aries-wm8994.yaml
@@ -0,0 +1,153 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung Aries audio complex with WM8994 codec
+
+maintainers:
+ - Jonathan Bakker <xc-racer2@live.ca>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ # With FM radio and modem master
+ - samsung,aries-wm8994
+ # Without FM radio and modem slave
+ - samsung,fascinate4g-wm8994
+
+ cpu:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 2
+ maxItems: 2
+ description: |
+ phandles to the I2S controller and bluetooth codec,
+ in that order
+ required:
+ - sound-dai
+
+ codec:
+ additionalProperties: false
+ type: object
+ properties:
+ sound-dai:
+ maxItems: 1
+ description: phandle to the WM8994 CODEC
+ required:
+ - sound-dai
+
+ samsung,audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ deprecated: true
+ description: |
+ List of the connections between audio
+ components; each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's source;
+ valid names for sources and sinks are the WM8994's pins (as
+ documented in its binding), and the jacks on the board -
+ For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic,
+ or FM In
+ For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic,
+ or HeadsetMic
+ Deprecated, use audio-routing.
+
+ extcon:
+ description: Extcon phandle for dock detection
+
+ main-micbias-supply:
+ description: Supply for the micbias on the main mic
+
+ headset-micbias-supply:
+ description: Supply for the micbias on the headset mic
+
+ earpath-sel-gpios:
+ maxItems: 1
+ description: GPIO for switching between tv-out and mic paths
+
+ headset-detect-gpios:
+ maxItems: 1
+ description: GPIO for detection of headset insertion
+
+ headset-key-gpios:
+ maxItems: 1
+ description: GPIO for detection of headset key press
+
+ io-channels:
+ maxItems: 1
+ description: IO channel to read micbias voltage for headset detection
+
+ io-channel-names:
+ const: headset-detect
+
+required:
+ - compatible
+ - cpu
+ - codec
+ - audio-routing
+ - extcon
+ - main-micbias-supply
+ - headset-micbias-supply
+ - earpath-sel-gpios
+ - headset-detect-gpios
+ - headset-key-gpios
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ sound {
+ compatible = "samsung,fascinate4g-wm8994";
+
+ model = "Fascinate4G";
+
+ extcon = <&fsa9480>;
+
+ main-micbias-supply = <&main_micbias_reg>;
+ headset-micbias-supply = <&headset_micbias_reg>;
+
+ earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>;
+
+ io-channels = <&adc 3>;
+ io-channel-names = "headset-detect";
+ headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>;
+ headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>;
+
+ audio-routing =
+ "HP", "HPOUT1L",
+ "HP", "HPOUT1R",
+
+ "SPK", "SPKOUTLN",
+ "SPK", "SPKOUTLP",
+
+ "RCV", "HPOUT2N",
+ "RCV", "HPOUT2P",
+
+ "LINE", "LINEOUT2N",
+ "LINE", "LINEOUT2P",
+
+ "IN1LP", "Main Mic",
+ "IN1LN", "Main Mic",
+
+ "IN1RP", "Headset Mic",
+ "IN1RN", "Headset Mic";
+
+ pinctrl-names = "default";
+ pinctrl-0 = <&headset_det &earpath_sel>;
+
+ cpu {
+ sound-dai = <&i2s0>, <&bt_codec>;
+ };
+
+ codec {
+ sound-dai = <&wm8994>;
+ };
+ };
diff --git a/Bindings/sound/samsung,arndale.yaml b/Bindings/sound/samsung,arndale.yaml
new file mode 100644
index 0000000..9bc4585
--- /dev/null
+++ b/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.
+ $ref: /schemas/types.yaml#/definitions/string
+
+required:
+ - compatible
+ - samsung,audio-codec
+ - samsung,audio-cpu
+
+additionalProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "samsung,arndale-rt5631";
+ samsung,audio-cpu = <&i2s0>;
+ samsung,audio-codec = <&rt5631>;
+ };
diff --git a/Bindings/sound/samsung,midas-audio.yaml b/Bindings/sound/samsung,midas-audio.yaml
new file mode 100644
index 0000000..6ec80f5
--- /dev/null
+++ b/Bindings/sound/samsung,midas-audio.yaml
@@ -0,0 +1,112 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung Midas audio complex with WM1811 codec
+
+maintainers:
+ - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ const: samsung,midas-audio
+
+ cpu:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ maxItems: 1
+ description: phandle to the I2S controller
+ required:
+ - sound-dai
+
+ codec:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ maxItems: 1
+ description: phandle to the WM1811 CODEC
+ required:
+ - sound-dai
+
+ samsung,audio-routing:
+ deprecated: true
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ List of the connections between audio components; each entry is
+ a pair of strings, the first being the connection's sink, the second
+ being the connection's source; valid names for sources and sinks are
+ the WM1811's pins (as documented in its binding), and the jacks
+ on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic.
+ Deprecated, use audio-routing.
+
+ mic-bias-supply:
+ description: Supply for the micbias on the Main microphone
+
+ submic-bias-supply:
+ description: Supply for the micbias on the Sub microphone
+
+ fm-sel-gpios:
+ maxItems: 1
+ description: GPIO pin for FM selection
+
+ lineout-sel-gpios:
+ maxItems: 1
+ description: GPIO pin for line out selection
+
+required:
+ - compatible
+ - cpu
+ - codec
+ - audio-routing
+ - mic-bias-supply
+ - submic-bias-supply
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ sound {
+ compatible = "samsung,midas-audio";
+ model = "Midas";
+
+ fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>;
+
+ mic-bias-supply = <&mic_bias_reg>;
+ submic-bias-supply = <&submic_bias_reg>;
+
+ audio-routing =
+ "HP", "HPOUT1L",
+ "HP", "HPOUT1R",
+
+ "SPK", "SPKOUTLN",
+ "SPK", "SPKOUTLP",
+ "SPK", "SPKOUTRN",
+ "SPK", "SPKOUTRP",
+
+ "RCV", "HPOUT2N",
+ "RCV", "HPOUT2P",
+
+ "IN1LP", "Main Mic",
+ "IN1LN", "Main Mic",
+ "IN1RP", "Sub Mic",
+ "IN1LP", "Sub Mic";
+
+ cpu {
+ sound-dai = <&i2s0>;
+ };
+
+ codec {
+ sound-dai = <&wm1811>;
+ };
+
+ };
diff --git a/Bindings/sound/samsung,odroid.yaml b/Bindings/sound/samsung,odroid.yaml
new file mode 100644
index 0000000..b77284e
--- /dev/null
+++ b/Bindings/sound/samsung,odroid.yaml
@@ -0,0 +1,97 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+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
+
+ assigned-clock-parents: true
+ assigned-clock-rates: true
+ assigned-clocks: true
+ clocks: true
+
+ cpu:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ description: phandles to the I2S controllers
+
+ codec:
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ items:
+ - description: phandle of the HDMI IP block node
+ - description: phandle of the MAX98090 CODEC
+
+ samsung,audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ deprecated: true
+ description: |
+ List of the connections between audio
+ components; each entry is a pair of strings, the first being the
+ 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
+ Deprecated, use audio-routing.
+
+ samsung,audio-widgets:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ This property specifies off-codec audio elements
+ like headphones or speakers, for details see widgets.txt
+
+required:
+ - compatible
+ - cpu
+ - codec
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "hardkernel,odroid-xu3-audio";
+ model = "Odroid-XU3";
+ audio-routing =
+ "Headphone Jack", "HPL",
+ "Headphone Jack", "HPR",
+ "IN1", "Mic Jack",
+ "Mic Jack", "MICBIAS";
+
+ cpu {
+ sound-dai = <&i2s0 0>;
+ };
+
+ codec {
+ sound-dai = <&hdmi>, <&max98090>;
+ };
+ };
diff --git a/Bindings/sound/samsung,smdk5250.yaml b/Bindings/sound/samsung,smdk5250.yaml
new file mode 100644
index 0000000..ac151d3
--- /dev/null
+++ b/Bindings/sound/samsung,smdk5250.yaml
@@ -0,0 +1,38 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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.
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+required:
+ - compatible
+ - samsung,audio-codec
+ - samsung,i2s-controller
+
+additionalProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "samsung,smdk-wm8994";
+ samsung,i2s-controller = <&i2s0>;
+ samsung,audio-codec = <&wm8994>;
+ };
diff --git a/Bindings/sound/samsung,snow.yaml b/Bindings/sound/samsung,snow.yaml
new file mode 100644
index 0000000..3d49aa4
--- /dev/null
+++ b/Bindings/sound/samsung,snow.yaml
@@ -0,0 +1,76 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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.
+ $ref: /schemas/types.yaml#/definitions/string
+
+required:
+ - compatible
+ - codec
+ - cpu
+
+additionalProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "google,snow-audio-max98095";
+ samsung,model = "Snow-I2S-MAX98095";
+
+ cpu {
+ sound-dai = <&i2s0 0>;
+ };
+
+ codec {
+ sound-dai = <&max98095 0>, <&hdmi>;
+ };
+ };
diff --git a/Bindings/sound/samsung,tm2.yaml b/Bindings/sound/samsung,tm2.yaml
new file mode 100644
index 0000000..7605925
--- /dev/null
+++ b/Bindings/sound/samsung,tm2.yaml
@@ -0,0 +1,80 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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>
+
+allOf:
+ - $ref: sound-card-common.yaml#
+
+properties:
+ compatible:
+ const: samsung,tm2-audio
+
+ audio-amplifier:
+ description: Phandle to the MAX98504 amplifier.
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ audio-codec:
+ description: Phandles to the codecs.
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ items:
+ - 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.
+ Deprecated, use audio-routing.
+ deprecated: true
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+
+ i2s-controller:
+ description: Phandles to the I2S controllers.
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ items:
+ - description: Phandle to I2S0.
+ - description: Phandle to I2S1.
+
+ mic-bias-gpios:
+ description: GPIO pin that enables the Main Mic bias regulator.
+
+required:
+ - compatible
+ - audio-amplifier
+ - audio-codec
+ - audio-routing
+ - i2s-controller
+ - mic-bias-gpios
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ sound {
+ compatible = "samsung,tm2-audio";
+ audio-codec = <&wm5110>, <&hdmi>;
+ i2s-controller = <&i2s0 0>, <&i2s1 0>;
+ audio-amplifier = <&max98504>;
+ mic-bias-gpios = <&gpr3 2 GPIO_ACTIVE_HIGH>;
+ model = "wm5110";
+ audio-routing = "HP", "HPOUT1L",
+ "HP", "HPOUT1R",
+ "SPK", "SPKOUT",
+ "SPKOUT", "HPOUT2L",
+ "SPKOUT", "HPOUT2R",
+ "RCV", "HPOUT3L",
+ "RCV", "HPOUT3R";
+ };
diff --git a/Bindings/sound/samsung-i2s.yaml b/Bindings/sound/samsung-i2s.yaml
new file mode 100644
index 0000000..30b3b6e
--- /dev/null
+++ b/Bindings/sound/samsung-i2s.yaml
@@ -0,0 +1,157 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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.
+
+ tesla,fsd-i2s: for 8/16/24bit stereo channel I2S for playback and
+ capture, secondary FIFO using external DMA, s/w reset control,
+ internal mux for root clock source with all root clock sampling
+ frequencies supported by Exynos7 I2S and 7.1 channel TDM support
+ for playback and capture TDM (Time division multiplexing) to allow
+ transfer of multiple channel audio data on single data line.
+ enum:
+ - samsung,s3c6410-i2s
+ - samsung,s5pv210-i2s
+ - samsung,exynos5420-i2s
+ - samsung,exynos7-i2s
+ - samsung,exynos7-i2s1
+ - tesla,fsd-i2s
+
+ '#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
+
+ 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:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: |
+ Internal DMA register base address of the audio
+ subsystem (used in secondary sound source).
+
+ power-domains:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+required:
+ - compatible
+ - reg
+ - dmas
+ - dma-names
+ - clocks
+ - clock-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/exynos-audss-clk.h>
+
+ i2s0: i2s@3830000 {
+ compatible = "samsung,s5pv210-i2s";
+ reg = <0x03830000 0x100>;
+ dmas = <&pdma0 10>,
+ <&pdma0 9>,
+ <&pdma0 8>;
+ dma-names = "tx", "rx", "tx-sec";
+ clocks = <&clock_audss EXYNOS_I2S_BUS>,
+ <&clock_audss EXYNOS_I2S_BUS>,
+ <&clock_audss EXYNOS_SCLK_I2S>;
+ clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
+ #clock-cells = <1>;
+ samsung,idma-addr = <0x03000000>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&i2s0_bus>;
+ #sound-dai-cells = <1>;
+ };
diff --git a/Bindings/sound/serial-midi.yaml b/Bindings/sound/serial-midi.yaml
new file mode 100644
index 0000000..f6a8073
--- /dev/null
+++ b/Bindings/sound/serial-midi.yaml
@@ -0,0 +1,51 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/serial-midi.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Generic Serial MIDI Interface
+
+maintainers:
+ - Daniel Kaehn <kaehndan@gmail.com>
+
+description:
+ Generic MIDI interface using a serial device. This denotes that a serial device is
+ dedicated to MIDI communication, either to an external MIDI device through a DIN5
+ or other connector, or to a known hardwired MIDI controller. This device must be a
+ child node of a serial node.
+
+ Can only be set to use standard baud rates corresponding to supported rates of the
+ parent serial device. If the standard MIDI baud of 31.25 kBaud is needed
+ (as would be the case if interfacing with arbitrary external MIDI devices),
+ configure the clocks of the parent serial device so that a requested baud of 38.4 kBaud
+ results in the standard MIDI baud rate, and set the 'current-speed' property to 38400 (default)
+
+properties:
+ compatible:
+ const: serial-midi
+
+ current-speed:
+ description: Baudrate to set the serial port to when this MIDI device is opened.
+ default: 38400
+
+required:
+ - compatible
+
+additionalProperties: false
+
+examples:
+ - |
+ serial {
+ midi {
+ compatible = "serial-midi";
+ };
+ };
+ - |
+ serial {
+ midi {
+ compatible = "serial-midi";
+ current-speed = <115200>;
+ };
+ };
diff --git a/Bindings/sound/sgtl5000.yaml b/Bindings/sound/sgtl5000.yaml
new file mode 100644
index 0000000..1353c05
--- /dev/null
+++ b/Bindings/sound/sgtl5000.yaml
@@ -0,0 +1,113 @@
+# SPDX-License-Identifier: GPL-2.0-only
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/sgtl5000.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale SGTL5000 Stereo Codec
+
+maintainers:
+ - Fabio Estevam <festevam@gmail.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: fsl,sgtl5000
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ assigned-clock-parents: true
+ assigned-clock-rates: true
+ assigned-clocks: true
+
+ clocks:
+ items:
+ - description: the clock provider of SYS_MCLK
+
+ VDDA-supply:
+ description: the regulator provider of VDDA
+
+ VDDIO-supply:
+ description: the regulator provider of VDDIO
+
+ VDDD-supply:
+ description: the regulator provider of VDDD
+
+ micbias-resistor-k-ohms:
+ description: The bias resistor to be used in kOhms. The resistor can take
+ values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not
+ mentioned or if the value is unknown, then micbias resistor is set to
+ 4k.
+ enum: [ 0, 2, 4, 8 ]
+
+ micbias-voltage-m-volts:
+ description: The bias voltage to be used in mVolts. The voltage can take
+ values from 1.25V to 3V by 250mV steps. If this node is not mentioned
+ or the value is unknown, then the value is set to 1.25V.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ]
+
+ lrclk-strength:
+ description: |
+ The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the
+ table below:
+
+ VDDIO 1.8V 2.5V 3.3V
+ 0 = Disable
+ 1 = 1.66 mA 2.87 mA 4.02 mA
+ 2 = 3.33 mA 5.74 mA 8.03 mA
+ 3 = 4.99 mA 8.61 mA 12.05 mA
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [ 0, 1, 2, 3 ]
+
+ sclk-strength:
+ description: |
+ The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the
+ table below:
+
+ VDDIO 1.8V 2.5V 3.3V
+ 0 = Disable
+ 1 = 1.66 mA 2.87 mA 4.02 mA
+ 2 = 3.33 mA 5.74 mA 8.03 mA
+ 3 = 4.99 mA 8.61 mA 12.05 mA
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [ 0, 1, 2, 3 ]
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+ - clocks
+ - VDDA-supply
+ - VDDIO-supply
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@a {
+ compatible = "fsl,sgtl5000";
+ reg = <0x0a>;
+ #sound-dai-cells = <0>;
+ clocks = <&clks 150>;
+ micbias-resistor-k-ohms = <2>;
+ micbias-voltage-m-volts = <2250>;
+ VDDA-supply = <®_3p3v>;
+ VDDIO-supply = <®_3p3v>;
+ };
+ };
+...
diff --git a/Bindings/sound/simple-audio-amplifier.yaml b/Bindings/sound/simple-audio-amplifier.yaml
new file mode 100644
index 0000000..5db1f98
--- /dev/null
+++ b/Bindings/sound/simple-audio-amplifier.yaml
@@ -0,0 +1,45 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/simple-audio-amplifier.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Simple Audio Amplifier
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - dioo,dio2125
+ - simple-audio-amplifier
+
+ enable-gpios:
+ maxItems: 1
+
+ VCC-supply:
+ description: >
+ power supply for the device
+
+ sound-name-prefix: true
+
+required:
+ - compatible
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/meson8-gpio.h>
+
+ analog-amplifier {
+ compatible = "simple-audio-amplifier";
+ VCC-supply = <®ulator>;
+ enable-gpios = <&gpio GPIOH_3 0>;
+ };
+
+...
diff --git a/Bindings/sound/simple-audio-mux.yaml b/Bindings/sound/simple-audio-mux.yaml
new file mode 100644
index 0000000..9f319ca
--- /dev/null
+++ b/Bindings/sound/simple-audio-mux.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/simple-audio-mux.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Simple Audio Multiplexer
+
+maintainers:
+ - Alexandre Belloni <aleandre.belloni@bootlin.com>
+
+description: |
+ Simple audio multiplexers are driven using gpios, allowing to select which of
+ their input line is connected to the output line.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: simple-audio-mux
+
+ mux-gpios:
+ description: |
+ GPIOs used to select the input line.
+
+ sound-name-prefix: true
+
+required:
+ - compatible
+ - mux-gpios
+
+additionalProperties: false
+
+examples:
+ - |
+ mux {
+ compatible = "simple-audio-mux";
+ mux-gpios = <&gpio 3 0>;
+ };
diff --git a/Bindings/sound/simple-card.yaml b/Bindings/sound/simple-card.yaml
new file mode 100644
index 0000000..59ac2d1
--- /dev/null
+++ b/Bindings/sound/simple-card.yaml
@@ -0,0 +1,559 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$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:
+ $ref: "#/definitions/system-clock-fixed"
+ required:
+ - sound-dai
+
+ additional-devs:
+ type: object
+ description:
+ Additional devices used by the simple audio card.
+ patternProperties:
+ '^iio-aux(-.+)?$':
+ type: object
+ $ref: audio-iio-aux.yaml#
+
+properties:
+ compatible:
+ contains:
+ 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,additional-devs:
+ $ref: "#/definitions/additional-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:
+ $ref: "#/definitions/pin-switches"
+ simple-audio-card,hp-det-gpio:
+ maxItems: 1
+ simple-audio-card,mic-det-gpio:
+ maxItems: 1
+
+patternProperties:
+ "^simple-audio-card,cpu(@[0-9a-f]+)?$":
+ $ref: "#/definitions/dai"
+ "^simple-audio-card,codec(@[0-9a-f]+)?$":
+ $ref: "#/definitions/dai"
+ "^simple-audio-card,plat(@[0-9a-f]+)?$":
+ $ref: "#/definitions/dai"
+
+ "^simple-audio-card,dai-link(@[0-9a-f]+)?$":
+ description: |
+ Container for dai-link level properties and the CPU and CODEC sub-nodes.
+ This container may be omitted when the card has only one DAI link.
+ type: object
+ properties:
+ reg:
+ maxItems: 1
+
+ "#address-cells":
+ const: 1
+ "#size-cells":
+ const: 0
+ # 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:
+ $ref: "#/definitions/pin-switches"
+ hp-det-gpio:
+ maxItems: 1
+ mic-det-gpio:
+ maxItems: 1
+
+ patternProperties:
+ "^cpu(-[0-9]+)?$":
+ $ref: "#/definitions/dai"
+ "^codec(-[0-9]+)?$":
+ $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>;
+ };
+ };
+
+# --------------------
+# route audio to/from a codec through an amplifier
+# designed with a potentiometer driven by IIO:
+# --------------------
+ - |
+ sound {
+ compatible = "simple-audio-card";
+
+ simple-audio-card,aux-devs = <&_in>, <&_out>;
+ simple-audio-card,routing =
+ "CODEC LEFTIN", "AMP_IN LEFT OUT",
+ "CODEC RIGHTIN", "AMP_IN RIGHT OUT",
+ "AMP_OUT LEFT IN", "CODEC LEFTOUT",
+ "AMP_OUT RIGHT IN", "CODEC RIGHTOUT";
+
+ simple-audio-card,additional-devs {
+ amp_out: iio-aux-out {
+ compatible = "audio-iio-aux";
+ io-channels = <&pot_out 0>, <&pot_out 1>;
+ io-channel-names = "LEFT", "RIGHT";
+ snd-control-invert-range = <1 1>;
+ sound-name-prefix = "AMP_OUT";
+ };
+
+ amp_in: iio_aux-in {
+ compatible = "audio-iio-aux";
+ io-channels = <&pot_in 0>, <&pot_in 1>;
+ io-channel-names = "LEFT", "RIGHT";
+ sound-name-prefix = "AMP_IN";
+ };
+ };
+
+ simple-audio-card,cpu {
+ sound-dai = <&cpu>;
+ };
+
+ simple-audio-card,codec {
+ sound-dai = <&codec>;
+ clocks = <&clocks>;
+ };
+ };
+
+# --------------------
+# Sampling Rate Conversion
+# --------------------
+ - |
+ sound {
+ compatible = "simple-audio-card";
+
+ 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: cpu-0 {
+ sound-dai = <&rcar_sound 1>;
+ };
+ cpu-1 {
+ sound-dai = <&rcar_sound 2>;
+ };
+ cpu-2 {
+ sound-dai = <&rcar_sound 3>;
+ };
+ cpu-3 {
+ sound-dai = <&rcar_sound 4>;
+ };
+ codec {
+ mclk-fs = <512>;
+ prefix = "pcm3168a";
+ dai-tdm-slot-num = <8>;
+ sound-dai = <&pcm3168a 0>;
+ };
+ };
+
+ simple-audio-card,dai-link@2 {
+ 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/Bindings/sound/sirf-audio-port.txt b/Bindings/sound/sirf-audio-port.txt
new file mode 100644
index 0000000..1f66de3
--- /dev/null
+++ b/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/Bindings/sound/sirf-audio.txt b/Bindings/sound/sirf-audio.txt
new file mode 100644
index 0000000..c88882c
--- /dev/null
+++ b/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/Bindings/sound/snps,designware-i2s.yaml b/Bindings/sound/snps,designware-i2s.yaml
new file mode 100644
index 0000000..a48d040
--- /dev/null
+++ b/Bindings/sound/snps,designware-i2s.yaml
@@ -0,0 +1,197 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/snps,designware-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: DesignWare I2S controller
+
+maintainers:
+ - Jose Abreu <joabreu@synopsys.com>
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - const: canaan,k210-i2s
+ - const: snps,designware-i2s
+ - enum:
+ - snps,designware-i2s
+ - starfive,jh7110-i2stx0
+ - starfive,jh7110-i2stx1
+ - starfive,jh7110-i2srx
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ description: |
+ The interrupt line number for the I2S controller. Add this
+ parameter if the I2S controller that you are using does not
+ support DMA.
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Sampling rate reference clock
+ - description: APB clock
+ - description: Audio master clock
+ - description: Inner audio master clock source
+ - description: External audio master clock source
+ - description: Bit clock
+ - description: Left/right channel clock
+ - description: External bit clock
+ - description: External left/right channel clock
+ minItems: 1
+
+ clock-names:
+ items:
+ - const: i2sclk
+ - const: apb
+ - const: mclk
+ - const: mclk_inner
+ - const: mclk_ext
+ - const: bclk
+ - const: lrck
+ - const: bclk_ext
+ - const: lrck_ext
+ minItems: 1
+
+ resets:
+ items:
+ - description: Optional controller resets
+ - description: controller reset of Sampling rate
+ minItems: 1
+
+ dmas:
+ items:
+ - description: TX DMA Channel
+ - description: RX DMA Channel
+ minItems: 1
+
+ dma-names:
+ items:
+ - const: tx
+ - const: rx
+ minItems: 1
+
+ starfive,syscon:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ items:
+ - items:
+ - description: phandle to System Register Controller sys_syscon node.
+ - description: I2S-rx enabled control offset of SYS_SYSCONSAIF__SYSCFG register.
+ - description: I2S-rx enabled control mask
+ description:
+ The phandle to System Register Controller syscon node and the I2S-rx(ADC)
+ enabled control offset and mask of SYS_SYSCONSAIF__SYSCFG register.
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: canaan,k210-i2s
+ then:
+ properties:
+ "#sound-dai-cells":
+ const: 1
+ else:
+ properties:
+ "#sound-dai-cells":
+ const: 0
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: snps,designware-i2s
+ then:
+ properties:
+ clocks:
+ maxItems: 1
+ clock-names:
+ maxItems: 1
+ resets:
+ maxItems: 1
+ else:
+ properties:
+ resets:
+ minItems: 2
+ maxItems: 2
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: starfive,jh7110-i2stx0
+ then:
+ properties:
+ clocks:
+ minItems: 5
+ maxItems: 5
+ clock-names:
+ minItems: 5
+ maxItems: 5
+ required:
+ - resets
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: starfive,jh7110-i2stx1
+ then:
+ properties:
+ clocks:
+ minItems: 9
+ maxItems: 9
+ clock-names:
+ minItems: 9
+ maxItems: 9
+ required:
+ - resets
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: starfive,jh7110-i2srx
+ then:
+ properties:
+ clocks:
+ minItems: 9
+ maxItems: 9
+ clock-names:
+ minItems: 9
+ maxItems: 9
+ required:
+ - resets
+ - starfive,syscon
+ else:
+ properties:
+ starfive,syscon: false
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+
+oneOf:
+ - required:
+ - dmas
+ - dma-names
+ - required:
+ - interrupts
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ soc_i2s: i2s@7ff90000 {
+ compatible = "snps,designware-i2s";
+ reg = <0x7ff90000 0x1000>;
+ clocks = <&scpi_i2sclk 0>;
+ clock-names = "i2sclk";
+ #sound-dai-cells = <0>;
+ dmas = <&dma0 5>;
+ dma-names = "tx";
+ };
diff --git a/Bindings/sound/soc-ac97link.txt b/Bindings/sound/soc-ac97link.txt
new file mode 100644
index 0000000..80152a8
--- /dev/null
+++ b/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/Bindings/sound/socionext,uniphier-aio.yaml b/Bindings/sound/socionext,uniphier-aio.yaml
new file mode 100644
index 0000000..8600520
--- /dev/null
+++ b/Bindings/sound/socionext,uniphier-aio.yaml
@@ -0,0 +1,102 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: UniPhier AIO audio system
+
+maintainers:
+ - <alsa-devel@alsa-project.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - socionext,uniphier-ld11-aio
+ - socionext,uniphier-ld20-aio
+ - socionext,uniphier-pxs2-aio
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clock-names:
+ const: aio
+
+ clocks:
+ maxItems: 1
+
+ reset-names:
+ const: aio
+
+ resets:
+ maxItems: 1
+
+ socionext,syscon:
+ description: |
+ Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF
+ signal pin to output from Hi-Z. This property is optional if you use I2S
+ signal pins only.
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ "#sound-dai-cells":
+ const: 1
+
+patternProperties:
+ "^port@[0-9]$":
+ description: |
+ Port number of DT node is specified by the following DAI channels that
+ depends on SoC.
+ ld11-aio,ld20-aio:
+ 0: hdmi
+ 1: pcmin2
+ 2: line
+ 3: hpcmout1
+ 4: pcmout3
+ 5: hiecout1
+ 6: epcmout2
+ 7: epcmout3
+ 8: hieccompout1
+ pxs2-aio:
+ 0: hdmi
+ 1: line
+ 2: aux
+ 3: hiecout1
+ 4: iecout1
+ 5: hieccompout1
+ 6: ieccompout1
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clock-names
+ - clocks
+ - reset-names
+ - resets
+ - "#sound-dai-cells"
+
+examples:
+ - |
+ audio@56000000 {
+ compatible = "socionext,uniphier-ld20-aio";
+ reg = <0x56000000 0x80000>;
+ interrupts = <0 144 4>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_aout>;
+ clock-names = "aio";
+ clocks = <&sys_clk 40>;
+ reset-names = "aio";
+ resets = <&sys_rst 40>;
+ #sound-dai-cells = <1>;
+ socionext,syscon = <&soc_glue>;
+ };
diff --git a/Bindings/sound/socionext,uniphier-evea.yaml b/Bindings/sound/socionext,uniphier-evea.yaml
new file mode 100644
index 0000000..9852776
--- /dev/null
+++ b/Bindings/sound/socionext,uniphier-evea.yaml
@@ -0,0 +1,75 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: UniPhier EVEA SoC-internal sound codec
+
+maintainers:
+ - <alsa-devel@alsa-project.org>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: socionext,uniphier-evea
+
+ reg:
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: evea
+ - const: exiv
+
+ clocks:
+ minItems: 2
+ maxItems: 2
+
+ reset-names:
+ items:
+ - const: evea
+ - const: exiv
+ - const: adamv
+
+ resets:
+ minItems: 3
+ maxItems: 3
+
+ "#sound-dai-cells":
+ const: 1
+
+patternProperties:
+ "^port@[0-9]$":
+ description: |
+ Port number of DT node is specified by the following DAI channels.
+ 0: line1
+ 1: hp
+ 2: line2
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - clock-names
+ - clocks
+ - reset-names
+ - resets
+ - "#sound-dai-cells"
+
+examples:
+ - |
+ codec@57900000 {
+ compatible = "socionext,uniphier-evea";
+ reg = <0x57900000 0x1000>;
+ clock-names = "evea", "exiv";
+ clocks = <&sys_clk 41>, <&sys_clk 42>;
+ reset-names = "evea", "exiv", "adamv";
+ resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
+ #sound-dai-cells = <1>;
+ };
diff --git a/Bindings/sound/sound-card-common.yaml b/Bindings/sound/sound-card-common.yaml
new file mode 100644
index 0000000..3a94117
--- /dev/null
+++ b/Bindings/sound/sound-card-common.yaml
@@ -0,0 +1,27 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/sound-card-common.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Board Sound Card Common Properties
+
+maintainers:
+ - Mark Brown <broonie@kernel.org>
+
+properties:
+ audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description: |
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+required:
+ - model
+
+additionalProperties: true
diff --git a/Bindings/sound/sound-dai.yaml b/Bindings/sound/sound-dai.yaml
new file mode 100644
index 0000000..ff9036e
--- /dev/null
+++ b/Bindings/sound/sound-dai.yaml
@@ -0,0 +1,20 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$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/Bindings/sound/spdif-receiver.txt b/Bindings/sound/spdif-receiver.txt
new file mode 100644
index 0000000..80f807b
--- /dev/null
+++ b/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/Bindings/sound/sprd-mcdt.txt b/Bindings/sound/sprd-mcdt.txt
new file mode 100644
index 0000000..274ba0a
--- /dev/null
+++ b/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/Bindings/sound/sprd-pcm.txt b/Bindings/sound/sprd-pcm.txt
new file mode 100644
index 0000000..fbbcade
--- /dev/null
+++ b/Bindings/sound/sprd-pcm.txt
@@ -0,0 +1,23 @@
+* Spreadtrum DMA platform 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/Bindings/sound/ssm4567.txt b/Bindings/sound/ssm4567.txt
new file mode 100644
index 0000000..ec3d9e7
--- /dev/null
+++ b/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/Bindings/sound/st,sta32x.txt b/Bindings/sound/st,sta32x.txt
new file mode 100644
index 0000000..52265fb
--- /dev/null
+++ b/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/Bindings/sound/st,sta350.txt b/Bindings/sound/st,sta350.txt
new file mode 100644
index 0000000..307398e
--- /dev/null
+++ b/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/Bindings/sound/st,sti-asoc-card.txt b/Bindings/sound/st,sti-asoc-card.txt
new file mode 100644
index 0000000..a6ffcde
--- /dev/null
+++ b/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/Bindings/sound/st,stm32-i2s.yaml b/Bindings/sound/st,stm32-i2s.yaml
new file mode 100644
index 0000000..b9111d3
--- /dev/null
+++ b/Bindings/sound/st,stm32-i2s.yaml
@@ -0,0 +1,105 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+ - reg
+ - clocks
+ - clock-names
+ - interrupts
+ - dmas
+ - dma-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/stm32mp1-clks.h>
+ i2s2: audio-controller@4000b000 {
+ compatible = "st,stm32h7-i2s";
+ #sound-dai-cells = <0>;
+ reg = <0x4000b000 0x400>;
+ clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>;
+ clock-names = "pclk", "i2sclk", "x8k", "x11k";
+ interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&dmamux1 39 0x400 0x01>,
+ <&dmamux1 40 0x400 0x01>;
+ dma-names = "rx", "tx";
+ pinctrl-names = "default";
+ pinctrl-0 = <&i2s2_pins_a>;
+
+ /* assume audio-graph */
+ port {
+ codec_endpoint: endpoint {
+ remote-endpoint = <&codec_endpoint>;
+ };
+ };
+ };
+
+...
diff --git a/Bindings/sound/st,stm32-sai.yaml b/Bindings/sound/st,stm32-sai.yaml
new file mode 100644
index 0000000..59df8a8
--- /dev/null
+++ b/Bindings/sound/st,stm32-sai.yaml
@@ -0,0 +1,199 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: STMicroelectronics STM32 Serial Audio Interface (SAI)
+
+maintainers:
+ - Olivier Moysan <olivier.moysan@foss.st.com>
+
+description:
+ The SAI interface (Serial Audio Interface) offers a wide set of audio
+ protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97.
+ The SAI contains two independent audio sub-blocks. Each sub-block has
+ its own clock generator and I/O lines controller.
+
+properties:
+ compatible:
+ enum:
+ - st,stm32f4-sai
+ - st,stm32h7-sai
+
+ reg:
+ items:
+ - description: Base address and size of SAI common register set.
+ - description: Base address and size of SAI identification register set.
+ minItems: 1
+
+ ranges:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 1
+
+ clocks:
+ maxItems: 3
+
+ clock-names:
+ maxItems: 3
+
+required:
+ - compatible
+ - reg
+ - ranges
+ - "#address-cells"
+ - "#size-cells"
+ - clocks
+ - clock-names
+
+patternProperties:
+ "^audio-controller@[0-9a-f]+$":
+ type: object
+ additionalProperties: false
+ description:
+ Two subnodes corresponding to SAI sub-block instances A et B
+ can be defined. Subnode can be omitted for unused sub-block.
+
+ properties:
+ compatible:
+ description: Compatible for SAI sub-block A or B.
+ pattern: "st,stm32-sai-sub-[ab]"
+
+ "#sound-dai-cells":
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: sai_ck clock feeding the internal clock generator.
+ - description: MCLK clock from a SAI set as master clock provider.
+ minItems: 1
+
+ clock-names:
+ items:
+ - const: sai_ck
+ - const: MCLK
+ minItems: 1
+
+ dmas:
+ maxItems: 1
+
+ dma-names:
+ description: |
+ rx: SAI sub-block is configured as a capture DAI.
+ tx: SAI sub-block is configured as a playback DAI.
+ enum: [ rx, tx ]
+
+ st,sync:
+ description:
+ Configure the SAI sub-block as slave of another SAI sub-block.
+ By default SAI sub-block is in asynchronous mode.
+ Must contain the phandle and index of the SAI sub-block providing
+ the synchronization.
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ items:
+ - items:
+ - description: phandle of the SAI sub-block
+ - description: index of the SAI sub-block
+
+ st,iec60958:
+ description:
+ If set, support S/PDIF IEC6958 protocol for playback.
+ IEC60958 protocol is not available for capture.
+ By default, custom protocol is assumed, meaning that protocol is
+ configured according to protocol defined in related DAI link node,
+ such as i2s, left justified, right justified, dsp and pdm protocols.
+ $ref: /schemas/types.yaml#/definitions/flag
+
+ "#clock-cells":
+ description: Configure the SAI device as master clock provider.
+ const: 0
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+ required:
+ - compatible
+ - "#sound-dai-cells"
+ - reg
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+allOf:
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: st,stm32f4-sai
+ then:
+ properties:
+ clocks:
+ items:
+ - description: x8k, SAI parent clock for sampling rates multiple of 8kHz.
+ - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz.
+
+ clock-names:
+ items:
+ - const: x8k
+ - const: x11k
+ else:
+ properties:
+ clocks:
+ items:
+ - description: pclk feeds the peripheral bus interface.
+ - description: x8k, SAI parent clock for sampling rates multiple of 8kHz.
+ - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz.
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: x8k
+ - const: x11k
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/stm32mp1-clks.h>
+ #include <dt-bindings/reset/stm32mp1-resets.h>
+ sai2: sai@4400b000 {
+ compatible = "st,stm32h7-sai";
+ #address-cells = <1>;
+ #size-cells = <1>;
+ ranges = <0 0x4400b000 0x400>;
+ reg = <0x4400b000 0x4>, <0x4400b3f0 0x10>;
+ clocks = <&rcc SAI2>, <&rcc PLL3_Q>, <&rcc PLL3_R>;
+ clock-names = "pclk", "x8k", "x11k";
+ pinctrl-names = "default", "sleep";
+ pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>;
+ pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>;
+
+ sai2a: audio-controller@4400b004 {
+ #sound-dai-cells = <0>;
+ compatible = "st,stm32-sai-sub-a";
+ reg = <0x4 0x1c>;
+ dmas = <&dmamux1 89 0x400 0x01>;
+ dma-names = "tx";
+ clocks = <&rcc SAI2_K>;
+ clock-names = "sai_ck";
+ };
+ };
+
+...
diff --git a/Bindings/sound/st,stm32-spdifrx.yaml b/Bindings/sound/st,stm32-spdifrx.yaml
new file mode 100644
index 0000000..bc48151
--- /dev/null
+++ b/Bindings/sound/st,stm32-spdifrx.yaml
@@ -0,0 +1,83 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - st,stm32h7-spdifrx
+
+ "#sound-dai-cells":
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: kclk
+
+ interrupts:
+ maxItems: 1
+
+ dmas:
+ items:
+ - description: audio data capture DMA
+ - description: IEC status bits capture DMA
+
+ dma-names:
+ items:
+ - const: rx
+ - const: rx-ctrl
+
+ resets:
+ maxItems: 1
+
+required:
+ - compatible
+ - "#sound-dai-cells"
+ - reg
+ - clocks
+ - clock-names
+ - interrupts
+ - dmas
+ - dma-names
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/stm32mp1-clks.h>
+ spdifrx: spdifrx@40004000 {
+ compatible = "st,stm32h7-spdifrx";
+ #sound-dai-cells = <0>;
+ reg = <0x40004000 0x400>;
+ clocks = <&rcc SPDIF_K>;
+ clock-names = "kclk";
+ interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>;
+ dmas = <&dmamux1 2 93 0x400 0x0>,
+ <&dmamux1 3 94 0x400 0x0>;
+ dma-names = "rx", "rx-ctrl";
+ pinctrl-0 = <&spdifrx_pins>;
+ pinctrl-names = "default";
+ };
+
+...
diff --git a/Bindings/sound/starfive,jh7110-pwmdac.yaml b/Bindings/sound/starfive,jh7110-pwmdac.yaml
new file mode 100644
index 0000000..e2b4db6
--- /dev/null
+++ b/Bindings/sound/starfive,jh7110-pwmdac.yaml
@@ -0,0 +1,76 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/starfive,jh7110-pwmdac.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: StarFive JH7110 PWM-DAC Controller
+
+description:
+ The PWM-DAC Controller uses PWM square wave generators plus RC filters to
+ form a DAC for audio play in StarFive JH7110 SoC. This audio play controller
+ supports 16 bit audio format, up to 48K sampling frequency, up to left and
+ right dual channels.
+
+maintainers:
+ - Hal Feng <hal.feng@starfivetech.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: starfive,jh7110-pwmdac
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: PWMDAC APB
+ - description: PWMDAC CORE
+
+ clock-names:
+ items:
+ - const: apb
+ - const: core
+
+ resets:
+ maxItems: 1
+ description: PWMDAC APB
+
+ dmas:
+ maxItems: 1
+ description: TX DMA Channel
+
+ dma-names:
+ const: tx
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - resets
+ - dmas
+ - dma-names
+ - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ pwmdac@100b0000 {
+ compatible = "starfive,jh7110-pwmdac";
+ reg = <0x100b0000 0x1000>;
+ clocks = <&syscrg 157>,
+ <&syscrg 158>;
+ clock-names = "apb", "core";
+ resets = <&syscrg 96>;
+ dmas = <&dma 22>;
+ dma-names = "tx";
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/starfive,jh7110-tdm.yaml b/Bindings/sound/starfive,jh7110-tdm.yaml
new file mode 100644
index 0000000..abb373f
--- /dev/null
+++ b/Bindings/sound/starfive,jh7110-tdm.yaml
@@ -0,0 +1,98 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/starfive,jh7110-tdm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: StarFive JH7110 TDM Controller
+
+description: |
+ The TDM Controller is a Time Division Multiplexed audio interface
+ integrated in StarFive JH7110 SoC, allowing up to 8 channels of
+ audio over a serial interface. The TDM controller can operate both
+ in master and slave mode.
+
+maintainers:
+ - Walker Chen <walker.chen@starfivetech.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - starfive,jh7110-tdm
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: TDM AHB Clock
+ - description: TDM APB Clock
+ - description: TDM Internal Clock
+ - description: TDM Clock
+ - description: Inner MCLK
+ - description: TDM External Clock
+
+ clock-names:
+ items:
+ - const: tdm_ahb
+ - const: tdm_apb
+ - const: tdm_internal
+ - const: tdm
+ - const: mclk_inner
+ - const: tdm_ext
+
+ resets:
+ items:
+ - description: tdm ahb reset line
+ - description: tdm apb reset line
+ - description: tdm core reset line
+
+ dmas:
+ items:
+ - description: RX DMA Channel
+ - description: TX DMA Channel
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - resets
+ - dmas
+ - dma-names
+ - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ tdm@10090000 {
+ compatible = "starfive,jh7110-tdm";
+ reg = <0x10090000 0x1000>;
+ clocks = <&syscrg 184>,
+ <&syscrg 185>,
+ <&syscrg 186>,
+ <&syscrg 187>,
+ <&syscrg 17>,
+ <&tdm_ext>;
+ clock-names = "tdm_ahb", "tdm_apb",
+ "tdm_internal", "tdm",
+ "mclk_inner", "tdm_ext";
+ resets = <&syscrg 105>,
+ <&syscrg 107>,
+ <&syscrg 106>;
+ dmas = <&dma 20>, <&dma 21>;
+ dma-names = "rx","tx";
+ #sound-dai-cells = <0>;
+ };
diff --git a/Bindings/sound/storm.txt b/Bindings/sound/storm.txt
new file mode 100644
index 0000000..062a4c1
--- /dev/null
+++ b/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/Bindings/sound/tas2552.txt b/Bindings/sound/tas2552.txt
new file mode 100644
index 0000000..a7eecad
--- /dev/null
+++ b/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/Bindings/sound/tas2562.yaml b/Bindings/sound/tas2562.yaml
new file mode 100644
index 0000000..f01c0dd
--- /dev/null
+++ b/Bindings/sound/tas2562.yaml
@@ -0,0 +1,83 @@
+# 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
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: TDM TX current sense time slot.
+
+ '#sound-dai-cells':
+ # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
+ # compatibility but is deprecated.
+ enum: [0, 1]
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec: codec@4c {
+ compatible = "ti,tas2562";
+ reg = <0x4c>;
+ #sound-dai-cells = <0>;
+ interrupt-parent = <&gpio1>;
+ interrupts = <14>;
+ shutdown-gpios = <&gpio1 15 0>;
+ ti,imon-slot-no = <0>;
+ };
+ };
diff --git a/Bindings/sound/tas2770.yaml b/Bindings/sound/tas2770.yaml
new file mode 100644
index 0000000..be2536e
--- /dev/null
+++ b/Bindings/sound/tas2770.yaml
@@ -0,0 +1,87 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2019-20 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/tas2770.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TAS2770 Smart PA
+
+maintainers:
+ - Shi Fu <shifu0704@thundersoft.com>
+
+description: |
+ The TAS2770 is a mono, digital input Class-D audio amplifier optimized for
+ efficiently driving high peak power into small loudspeakers.
+ Integrated speaker voltage and current sense provides for
+ real time monitoring of loudspeaker behavior.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - ti,tas2770
+
+ reg:
+ maxItems: 1
+ description: |
+ I2C address of the device can be between 0x41 to 0x48.
+
+ reset-gpio:
+ maxItems: 1
+ description: GPIO used to reset the device.
+
+ shutdown-gpios:
+ maxItems: 1
+ description: GPIO used to control the state of the device.
+
+ interrupts:
+ maxItems: 1
+
+ ti,imon-slot-no:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: TDM TX current sense time slot.
+
+ ti,vmon-slot-no:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: TDM TX voltage sense time slot.
+
+ ti,asi-format:
+ deprecated: true
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: Sets TDM RX capture edge.
+ enum:
+ - 0 # Rising edge
+ - 1 # Falling edge
+
+ '#sound-dai-cells':
+ # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
+ # compatibility but is deprecated.
+ enum: [0, 1]
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec: codec@41 {
+ compatible = "ti,tas2770";
+ reg = <0x41>;
+ #sound-dai-cells = <0>;
+ interrupt-parent = <&gpio1>;
+ interrupts = <14>;
+ reset-gpio = <&gpio1 15 0>;
+ shutdown-gpios = <&gpio1 14 0>;
+ ti,imon-slot-no = <0>;
+ ti,vmon-slot-no = <2>;
+ };
+ };
diff --git a/Bindings/sound/tas27xx.yaml b/Bindings/sound/tas27xx.yaml
new file mode 100644
index 0000000..f2d878f
--- /dev/null
+++ b/Bindings/sound/tas27xx.yaml
@@ -0,0 +1,82 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020-2022 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/tas27xx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TAS2764/TAS2780 Smart PA
+
+maintainers:
+ - Shenghao Ding <shenghao-ding@ti.com>
+
+description: |
+ The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier
+ optimized for efficiently driving high peak power into small
+ loudspeakers. Integrated speaker voltage and current sense provides
+ for real time monitoring of loudspeaker behavior.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - ti,tas2764
+ - ti,tas2780
+
+ reg:
+ maxItems: 1
+ description: |
+ I2C address of the device can be between 0x38 to 0x45.
+
+ reset-gpios:
+ maxItems: 1
+ description: GPIO used to reset the device.
+
+ shutdown-gpios:
+ maxItems: 1
+ description: GPIO used to control the state of the device.
+
+ interrupts:
+ maxItems: 1
+
+ ti,imon-slot-no:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: TDM TX current sense time slot.
+
+ ti,vmon-slot-no:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: TDM TX voltage sense time slot.
+
+ '#sound-dai-cells':
+ # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
+ # compatibility but is deprecated.
+ enum: [0, 1]
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec: codec@38 {
+ compatible = "ti,tas2764";
+ reg = <0x38>;
+ #sound-dai-cells = <0>;
+ interrupt-parent = <&gpio1>;
+ interrupts = <14>;
+ reset-gpios = <&gpio1 15 0>;
+ shutdown-gpios = <&gpio1 15 0>;
+ ti,imon-slot-no = <0>;
+ ti,vmon-slot-no = <2>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/tas571x.txt b/Bindings/sound/tas571x.txt
new file mode 100644
index 0000000..1addc75
--- /dev/null
+++ b/Bindings/sound/tas571x.txt
@@ -0,0 +1,49 @@
+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"
+ - "ti,tas5733"
+- 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/Bindings/sound/tas5720.txt b/Bindings/sound/tas5720.txt
new file mode 100644
index 0000000..7d851ae
--- /dev/null
+++ b/Bindings/sound/tas5720.txt
@@ -0,0 +1,28 @@
+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/TAS5720A-Q1
+https://www.ti.com/product/TAS5722L
+
+Required properties:
+
+- compatible : "ti,tas5720",
+ "ti,tas5720a-q1",
+ "ti,tas5722"
+- reg : I2C slave address
+- dvdd-supply : phandle to a 3.3-V supply for the digital circuitry
+- 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/Bindings/sound/tas5805m.yaml b/Bindings/sound/tas5805m.yaml
new file mode 100644
index 0000000..12c4197
--- /dev/null
+++ b/Bindings/sound/tas5805m.yaml
@@ -0,0 +1,57 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/tas5805m.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: TAS5805M audio amplifier
+
+maintainers:
+ - Daniel Beer <daniel.beer@igorinstitute.com>
+
+description: |
+ The TAS5805M is a class D audio amplifier with a built-in DSP.
+
+properties:
+ compatible:
+ enum:
+ - ti,tas5805m
+
+ reg:
+ maxItems: 1
+ description: |
+ I2C address of the amplifier. See the datasheet for possible values.
+
+ pvdd-supply:
+ description: |
+ Regulator for audio power supply (PVDD in the datasheet).
+
+ pdn-gpios:
+ description: |
+ Power-down control GPIO (PDN pin in the datasheet).
+
+ ti,dsp-config-name:
+ description: |
+ The name of the DSP configuration that should be loaded for this
+ instance. Configuration blobs are sequences of register writes
+ generated from TI's PPC3 tool.
+ $ref: /schemas/types.yaml#/definitions/string
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ tas5805m: tas5805m@2c {
+ reg = <0x2c>;
+ compatible = "ti,tas5805m";
+
+ pvdd-supply = <&audiopwr>;
+ pdn-gpios = <&tlmm 160 0>;
+
+ ti,dsp-config-name = "mono_pbtl_48khz";
+ };
+ };
+...
diff --git a/Bindings/sound/tda7419.txt b/Bindings/sound/tda7419.txt
new file mode 100644
index 0000000..6b85ec3
--- /dev/null
+++ b/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/Bindings/sound/tdm-slot.txt b/Bindings/sound/tdm-slot.txt
new file mode 100644
index 0000000..4bb513a
--- /dev/null
+++ b/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/Bindings/sound/test-component.yaml b/Bindings/sound/test-component.yaml
new file mode 100644
index 0000000..9c40a21
--- /dev/null
+++ b/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#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Test Component
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+ compatible:
+ enum:
+ - test-cpu
+ - test-cpu-verbose
+ - test-cpu-verbose-dai
+ - test-cpu-verbose-component
+ - test-codec
+ - test-codec-verbose
+ - test-codec-verbose-dai
+ - test-codec-verbose-component
+
+required:
+ - compatible
+
+additionalProperties: true
+
+examples:
+ - |
+ test_cpu {
+ compatible = "test-cpu";
+ };
diff --git a/Bindings/sound/ti,ads117x.txt b/Bindings/sound/ti,ads117x.txt
new file mode 100644
index 0000000..7db19b5
--- /dev/null
+++ b/Bindings/sound/ti,ads117x.txt
@@ -0,0 +1,11 @@
+Texas Intstruments ADS117x ADC
+
+Required properties:
+
+ - compatible : "ti,ads1174" or "ti,ads1178"
+
+Example:
+
+ads1178 {
+ compatible = "ti,ads1178";
+};
diff --git a/Bindings/sound/ti,j721e-cpb-audio.yaml b/Bindings/sound/ti,j721e-cpb-audio.yaml
new file mode 100644
index 0000000..20ea588
--- /dev/null
+++ b/Bindings/sound/ti,j721e-cpb-audio.yaml
@@ -0,0 +1,139 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020 Texas Instruments Incorporated
+# Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+%YAML 1.2
+---
+$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/Bindings/sound/ti,j721e-cpb-ivi-audio.yaml b/Bindings/sound/ti,j721e-cpb-ivi-audio.yaml
new file mode 100644
index 0000000..5b2874a
--- /dev/null
+++ b/Bindings/sound/ti,j721e-cpb-ivi-audio.yaml
@@ -0,0 +1,145 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020 Texas Instruments Incorporated
+# Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments J721e Common Processor Board Audio Support
+
+maintainers:
+ - Peter Ujfalusi <peter.ujfalusi@gmail.com>
+
+description: |
+ The Infotainment board plugs into the Common Processor Board, the support of the
+ extension board is extending the CPB audio support, described in:
+ sound/ti,j721e-cpb-audio.txt
+
+ The audio support on the Infotainment Expansion Board consists of McASP0
+ connected to two pcm3168a codecs with dedicated set of serializers to each.
+ The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
+
+ In order to support 48KHz and 44.1KHz family of sampling rates the parent clock
+ for AUDIO_REFCLK0 needs to be changed between PLL4 (for 48KHz) and PLL15 (for
+ 44.1KHz). The same PLLs are used for McASP0's AUXCLK clock via different
+ HSDIVIDER.
+
+ Note: the same PLL4 and PLL15 is used by the audio support on the CPB!
+
+ Clocking setup for 48KHz family:
+ PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
+ | |-> MCASP0_AUXCLK ---> McASP0.auxclk
+ |
+ |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI
+ |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI
+
+ Clocking setup for 44.1KHz family:
+ PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
+ | |-> MCASP0_AUXCLK ---> McASP0.auxclk
+ |
+ |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI
+ |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI
+
+properties:
+ compatible:
+ items:
+ - const: ti,j721e-cpb-ivi-audio
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+ ti,cpb-mcasp:
+ description: phandle to McASP used on CPB
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ ti,cpb-codec:
+ description: phandle to the pcm3168a codec used on the CPB
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ ti,ivi-mcasp:
+ description: phandle to McASP used on IVI
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ ti,ivi-codec-a:
+ description: phandle to the pcm3168a-A codec on the expansion board
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ ti,ivi-codec-b:
+ description: phandle to the pcm3168a-B codec on the expansion board
+ $ref: /schemas/types.yaml#/definitions/phandle
+
+ clocks:
+ items:
+ - description: AUXCLK clock for McASP used by CPB audio
+ - description: Parent for CPB_McASP auxclk (for 48KHz)
+ - description: Parent for CPB_McASP auxclk (for 44.1KHz)
+ - description: SCKI clock for the pcm3168a codec on CPB
+ - description: Parent for CPB_SCKI clock (for 48KHz)
+ - description: Parent for CPB_SCKI clock (for 44.1KHz)
+ - description: AUXCLK clock for McASP used by IVI audio
+ - description: Parent for IVI_McASP auxclk (for 48KHz)
+ - description: Parent for IVI_McASP auxclk (for 44.1KHz)
+ - description: SCKI clock for the pcm3168a codec on IVI
+ - description: Parent for IVI_SCKI clock (for 48KHz)
+ - description: Parent for IVI_SCKI clock (for 44.1KHz)
+
+ clock-names:
+ items:
+ - const: cpb-mcasp-auxclk
+ - const: cpb-mcasp-auxclk-48000
+ - const: cpb-mcasp-auxclk-44100
+ - const: cpb-codec-scki
+ - const: cpb-codec-scki-48000
+ - const: cpb-codec-scki-44100
+ - const: ivi-mcasp-auxclk
+ - const: ivi-mcasp-auxclk-48000
+ - const: ivi-mcasp-auxclk-44100
+ - const: ivi-codec-scki
+ - const: ivi-codec-scki-48000
+ - const: ivi-codec-scki-44100
+
+required:
+ - compatible
+ - model
+ - ti,cpb-mcasp
+ - ti,cpb-codec
+ - ti,ivi-mcasp
+ - ti,ivi-codec-a
+ - ti,ivi-codec-b
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |+
+ sound {
+ compatible = "ti,j721e-cpb-ivi-audio";
+ model = "j721e-cpb-ivi";
+
+ ti,cpb-mcasp = <&mcasp10>;
+ ti,cpb-codec = <&pcm3168a_1>;
+
+ ti,ivi-mcasp = <&mcasp0>;
+ ti,ivi-codec-a = <&pcm3168a_a>;
+ ti,ivi-codec-b = <&pcm3168a_b>;
+
+ clocks = <&k3_clks 184 1>,
+ <&k3_clks 184 2>, <&k3_clks 184 4>,
+ <&k3_clks 157 371>,
+ <&k3_clks 157 400>, <&k3_clks 157 401>,
+ <&k3_clks 174 1>,
+ <&k3_clks 174 2>, <&k3_clks 174 4>,
+ <&k3_clks 157 301>,
+ <&k3_clks 157 330>, <&k3_clks 157 331>;
+ clock-names = "cpb-mcasp-auxclk",
+ "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100",
+ "cpb-codec-scki",
+ "cpb-codec-scki-48000", "cpb-codec-scki-44100",
+ "ivi-mcasp-auxclk",
+ "ivi-mcasp-auxclk-48000", "ivi-mcasp-auxclk-44100",
+ "ivi-codec-scki",
+ "ivi-codec-scki-48000", "ivi-codec-scki-44100";
+ };
diff --git a/Bindings/sound/ti,pcm1681.txt b/Bindings/sound/ti,pcm1681.txt
new file mode 100644
index 0000000..4df1718
--- /dev/null
+++ b/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/Bindings/sound/ti,pcm3168a.yaml b/Bindings/sound/ti,pcm3168a.yaml
new file mode 100644
index 0000000..0b4f003
--- /dev/null
+++ b/Bindings/sound/ti,pcm3168a.yaml
@@ -0,0 +1,108 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,pcm3168a.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments PCM3168A Audio Codec
+
+maintainers:
+ - Damien Horsley <Damien.Horsley@imgtec.com>
+ - Geert Uytterhoeven <geert+renesas@glider.be>
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+description:
+ The Texas Instruments PCM3168A is a 24-bit Multi-channel Audio CODEC with
+ 96/192kHz sampling rate, supporting both SPI and I2C bus access.
+
+properties:
+ compatible:
+ const: ti,pcm3168a
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: System clock input
+
+ clock-names:
+ items:
+ - const: scki
+
+ reset-gpios:
+ items:
+ - description: |
+ GPIO line connected to the active-low RST pin of the codec.
+ RST = low: device power-down
+ RST = high: device is enabled
+
+ "#sound-dai-cells":
+ enum: [0, 1]
+
+ VDD1-supply:
+ description: Digital power supply regulator 1 (+3.3V)
+
+ VDD2-supply:
+ description: Digital power supply regulator 2 (+3.3V)
+
+ VCCAD1-supply:
+ description: ADC power supply regulator 1 (+5V)
+
+ VCCAD2-supply:
+ description: ADC power supply regulator 2 (+5V)
+
+ VCCDA1-supply:
+ description: DAC power supply regulator 1 (+5V)
+
+ VCCDA2-supply:
+ description: DAC power supply regulator 2 (+5V)
+
+ ports:
+ $ref: audio-graph-port.yaml#/definitions/port-base
+ unevaluatedProperties: false
+ properties:
+ port@0:
+ $ref: audio-graph-port.yaml#
+ description: Audio input port.
+
+ port@1:
+ $ref: audio-graph-port.yaml#
+ description: Audio output port.
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - VDD1-supply
+ - VDD2-supply
+ - VCCAD1-supply
+ - VCCAD2-supply
+ - VCCDA1-supply
+ - VCCDA2-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ pcm3168a: audio-codec@44 {
+ compatible = "ti,pcm3168a";
+ reg = <0x44>;
+ reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>;
+ clocks = <&clk_core 42>;
+ clock-names = "scki";
+ VDD1-supply = <&supply3v3>;
+ VDD2-supply = <&supply3v3>;
+ VCCAD1-supply = <&supply5v0>;
+ VCCAD2-supply = <&supply5v0>;
+ VCCDA1-supply = <&supply5v0>;
+ VCCDA2-supply = <&supply5v0>;
+ };
+ };
diff --git a/Bindings/sound/ti,src4xxx.yaml b/Bindings/sound/ti,src4xxx.yaml
new file mode 100644
index 0000000..27230c6
--- /dev/null
+++ b/Bindings/sound/ti,src4xxx.yaml
@@ -0,0 +1,48 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments SRC4392
+
+description: |
+ The SRC4392 is a digital audio codec that can be connected via
+ I2C or SPI. Currently, only I2C bus is supported.
+
+maintainers:
+ - Matt Flax <flatmax@flatmax.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: ti,src4392
+
+ "#sound-dai-cells":
+ const: 0
+
+ reg:
+ maxItems: 1
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@70 {
+ #sound-dai-cells = <0>;
+ compatible = "ti,src4392";
+ reg = <0x70>;
+ };
+ };
+...
diff --git a/Bindings/sound/ti,tas2781.yaml b/Bindings/sound/ti,tas2781.yaml
new file mode 100644
index 0000000..a69e6c2
--- /dev/null
+++ b/Bindings/sound/ti,tas2781.yaml
@@ -0,0 +1,74 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2022 - 2023 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TAS2781 SmartAMP
+
+maintainers:
+ - Shenghao Ding <shenghao-ding@ti.com>
+
+description:
+ The TAS2781 is a mono, digital input Class-D audio amplifier
+ optimized for efficiently driving high peak power into small
+ loudspeakers. An integrated on-chip DSP supports Texas Instruments
+ Smart Amp speaker protection algorithm. The integrated speaker
+ voltage and current sense provides for real time
+ monitoring of loudspeaker behavior.
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - ti,tas2781
+
+ reg:
+ description:
+ I2C address, in multiple tas2781s case, all the i2c address
+ aggregate as one Audio Device to support multiple audio slots.
+ maxItems: 8
+ minItems: 1
+ items:
+ minimum: 0x38
+ maximum: 0x3f
+
+ reset-gpios:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ /* example with quad tas2781s, such as tablet or pad device */
+ #address-cells = <1>;
+ #size-cells = <0>;
+ quad_tas2781: tas2781@38 {
+ compatible = "ti,tas2781";
+ reg = <0x38>, /* Audio slot 0 */
+ <0x3a>, /* Audio slot 1 */
+ <0x39>, /* Audio slot 2 */
+ <0x3b>; /* Audio slot 3 */
+
+ #sound-dai-cells = <0>;
+ reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>;
+ interrupt-parent = <&gpio1>;
+ interrupts = <15>;
+ };
+ };
+...
diff --git a/Bindings/sound/ti,tas5086.txt b/Bindings/sound/ti,tas5086.txt
new file mode 100644
index 0000000..234dad2
--- /dev/null
+++ b/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/Bindings/sound/ti,tas6424.txt b/Bindings/sound/ti,tas6424.txt
new file mode 100644
index 0000000..00940c4
--- /dev/null
+++ b/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/Bindings/sound/ti,tlv320adc3xxx.yaml b/Bindings/sound/ti,tlv320adc3xxx.yaml
new file mode 100644
index 0000000..ede14ca
--- /dev/null
+++ b/Bindings/sound/ti,tlv320adc3xxx.yaml
@@ -0,0 +1,140 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC
+
+maintainers:
+ - Ricard Wanderlof <ricardw@axis.com>
+
+description: |
+ Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC
+ https://www.ti.com/product/TLV320ADC3001
+ https://www.ti.com/product/TLV320ADC3101
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - ti,tlv320adc3001
+ - ti,tlv320adc3101
+
+ reg:
+ maxItems: 1
+ description: I2C address
+
+ '#sound-dai-cells':
+ const: 0
+
+ '#gpio-cells':
+ const: 2
+
+ gpio-controller: true
+
+ reset-gpios:
+ maxItems: 1
+ description: GPIO pin used for codec reset (RESET pin)
+
+ clocks:
+ maxItems: 1
+ description: Master clock (MCLK)
+
+ ti,dmdin-gpio1:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used
+ - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions
+ - 2 # ADC3XXX_GPIO_GPI - General purpose input
+ - 3 # ADC3XXX_GPIO_GPO - General purpose output
+ - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg
+ - 5 # ADC3XXX_GPIO_INT1 - INT1 output
+ - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK
+ - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK
+ default: 0
+ description: |
+ Configuration for DMDIN/GPIO1 pin.
+
+ When ADC3XXX_GPIO_GPO is configured, this causes corresponding the
+ ALSA control "GPIOx Output" to appear, as a switch control.
+
+ ti,dmclk-gpio2:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used
+ - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions
+ - 2 # ADC3XXX_GPIO_GPI - General purpose input
+ - 3 # ADC3XXX_GPIO_GPO - General purpose output
+ - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg
+ - 5 # ADC3XXX_GPIO_INT1 - INT1 output
+ - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK
+ - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK
+ default: 0
+ description: |
+ Configuration for DMCLK/GPIO2 pin.
+
+ When ADC3XXX_GPIO_GPO is configured, this causes corresponding the
+ ALSA control "GPIOx Output" to appear, as a switch control.
+
+ Note that there is currently no support for reading the GPIO pins as
+ inputs.
+
+ ti,micbias1-vg:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down
+ - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V
+ - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V
+ - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply
+ default: 0
+ description: |
+ Mic bias voltage output on MICBIAS1 pin
+
+ ti,micbias2-vg:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum:
+ - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down
+ - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V
+ - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V
+ - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply
+ default: 0
+ description: |
+ Mic bias voltage output on MICBIAS2 pin
+
+required:
+ - compatible
+ - reg
+ - clocks
+
+unevaluatedProperties: false
+
+examples:
+ - |
+
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/sound/tlv320adc3xxx.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ tlv320adc3101: audio-codec@18 {
+ compatible = "ti,tlv320adc3101";
+ reg = <0x18>;
+ reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>;
+ clocks = <&audio_mclk>;
+ gpio-controller;
+ #gpio-cells = <2>;
+ ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>;
+ ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>;
+ };
+ };
+
+ audio_mclk: clock {
+ compatible = "fixed-clock";
+ #clock-cells = <0>;
+ clock-frequency = <24576000>;
+ };
+...
diff --git a/Bindings/sound/ti,tlv320aic32x4.yaml b/Bindings/sound/ti,tlv320aic32x4.yaml
new file mode 100644
index 0000000..a7cc9aa
--- /dev/null
+++ b/Bindings/sound/ti,tlv320aic32x4.yaml
@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2019 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,tlv320aic32x4.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TLV320AIC32x4 Stereo Audio codec
+
+maintainers:
+ - Alexander Stein <alexander.stein@ew.tq-group.com>
+
+description: |
+ The TLV320AIC32x4 audio codec can be accessed using I2C or SPI
+
+properties:
+ compatible:
+ enum:
+ - ti,tas2505
+ - ti,tlv320aic32x4
+ - ti,tlv320aic32x6
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Master clock
+
+ clock-names:
+ items:
+ - const: mclk
+
+ av-supply:
+ description: Analog core power supply
+
+ dv-supply:
+ description: Digital core power supply
+
+ iov-supply:
+ description: Digital IO power supply
+
+ ldoin-supply:
+ description: LDO power supply
+
+ reset-gpios:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+ aic32x4-gpio-func:
+ description: |
+ GPIO function configuration for pins MFP1-MFP5.
+ Types are defined in include/sound/tlv320aic32x4.h
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 5
+ maxItems: 5
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - iov-supply
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ not:
+ required:
+ - ldoin-supply
+ then:
+ required:
+ - av-supply
+ - dv-supply
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@18 {
+ compatible = "ti,tlv320aic32x4";
+ reg = <0x18>;
+ iov-supply = <®_3v3>;
+ ldoin-supply = <®_3v3>;
+ clocks = <&clks 201>;
+ clock-names = "mclk";
+ aic32x4-gpio-func= <
+ 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+ 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+ 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */
+ 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+ 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */
+ >;
+ };
+ };
diff --git a/Bindings/sound/ti,tlv320aic3x.yaml b/Bindings/sound/ti,tlv320aic3x.yaml
new file mode 100644
index 0000000..206f6d6
--- /dev/null
+++ b/Bindings/sound/ti,tlv320aic3x.yaml
@@ -0,0 +1,166 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+# Copyright (C) 2022 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TLV320AIC3x Codec
+
+description: |
+ TLV320AIC3x are a series of low-power stereo audio codecs with stereo
+ headphone amplifier, as well as multiple inputs and outputs programmable in
+ single-ended or fully differential configurations.
+
+ The serial control bus supports SPI or I2C protocols, while the serial audio
+ data bus is programmable for I2S, left/right-justified, DSP, or TDM modes.
+
+ The following pins can be referred in the sound node's audio routing property:
+
+ CODEC output pins:
+ LLOUT
+ RLOUT
+ MONO_LOUT
+ HPLOUT
+ HPROUT
+ HPLCOM
+ HPRCOM
+
+ CODEC input pins for TLV320AIC3104:
+ MIC2L
+ MIC2R
+ LINE1L
+ LINE1R
+
+ CODEC input pins for other compatible codecs:
+ MIC3L
+ MIC3R
+ LINE1L
+ LINE2L
+ LINE1R
+ LINE2R
+
+maintainers:
+ - Jai Luthra <j-luthra@ti.com>
+
+properties:
+ compatible:
+ enum:
+ - ti,tlv320aic3x
+ - ti,tlv320aic33
+ - ti,tlv320aic3007
+ - ti,tlv320aic3106
+ - ti,tlv320aic3104
+
+ reg:
+ maxItems: 1
+
+ reset-gpios:
+ maxItems: 1
+ description:
+ GPIO specification for the active low RESET input.
+
+ gpio-reset:
+ $ref: /schemas/types.yaml#/definitions/uint32-matrix
+ maxItems: 1
+ description:
+ Deprecated, please use reset-gpios instead.
+ deprecated: true
+
+ ai3x-gpio-func:
+ description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ maxItems: 2
+
+ ai3x-micbias-vg:
+ description: MicBias required voltage. If node is omitted then MicBias is powered down.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ oneOf:
+ - const: 1
+ description: MICBIAS output is powered to 2.0V.
+ - const: 2
+ description: MICBIAS output is powered to 2.5V.
+ - const: 3
+ description: MICBIAS output is connected to AVDD.
+
+ ai3x-ocmv:
+ description: Output Common-Mode Voltage selection.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ oneOf:
+ - const: 0
+ description: 1.35V
+ - const: 1
+ description: 1.5V
+ - const: 2
+ description: 1.65V
+ - const: 3
+ description: 1.8V
+
+ AVDD-supply:
+ description: Analog DAC voltage.
+
+ IOVDD-supply:
+ description: I/O voltage.
+
+ DRVDD-supply:
+ description: ADC analog and output driver voltage.
+
+ DVDD-supply:
+ description: Digital core voltage.
+
+ '#sound-dai-cells':
+ const: 0
+
+ clocks:
+ maxItems: 1
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ tlv320aic3x_i2c: audio-codec@1b {
+ compatible = "ti,tlv320aic3x";
+ reg = <0x1b>;
+
+ reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
+ AVDD-supply = <®ulator>;
+ IOVDD-supply = <®ulator>;
+ DRVDD-supply = <®ulator>;
+ DVDD-supply = <®ulator>;
+ };
+ };
+
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ tlv320aic3x_spi: audio-codec@0 {
+ compatible = "ti,tlv320aic3x";
+ reg = <0>; /* CS number */
+ #sound-dai-cells = <0>;
+
+ AVDD-supply = <®ulator>;
+ IOVDD-supply = <®ulator>;
+ DRVDD-supply = <®ulator>;
+ DVDD-supply = <®ulator>;
+ ai3x-ocmv = <0>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/ti,ts3a227e.yaml b/Bindings/sound/ti,ts3a227e.yaml
new file mode 100644
index 0000000..7859306
--- /dev/null
+++ b/Bindings/sound/ti,ts3a227e.yaml
@@ -0,0 +1,94 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: Intended MICBIAS voltage (datasheet section 9.6.7).
+ enum:
+ - 0 # 2.1 V
+ - 1 # 2.2 V
+ - 2 # 2.3 V
+ - 3 # 2.4 V
+ - 4 # 2.5 V
+ - 5 # 2.6 V
+ - 6 # 2.7 V
+ - 7 # 2.8 V
+ default: 1
+
+ ti,debounce-release-ms:
+ description: key release debounce time in ms (datasheet section 9.6.7).
+ enum:
+ - 0
+ - 20
+ default: 20
+
+ ti,debounce-press-ms:
+ description: key press debounce time in ms (datasheet section 9.6.7).
+ enum:
+ - 2
+ - 40
+ - 80
+ - 120
+ default: 80
+
+ ti,debounce-insertion-ms:
+ description: headset insertion debounce time in ms (datasheet section 9.6.5).
+ enum:
+ - 2
+ - 30
+ - 60
+ - 90
+ - 120
+ - 150
+ - 1000
+ - 2000
+ default: 90
+
+required:
+ - compatible
+ - reg
+ - interrupts
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/irq.h>
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec: audio-controller@3b {
+ compatible = "ti,ts3a227e";
+ reg = <0x3b>;
+ interrupt-parent = <&gpio1>;
+ interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/tlv320adcx140.yaml b/Bindings/sound/tlv320adcx140.yaml
new file mode 100644
index 0000000..f3274bc
--- /dev/null
+++ b/Bindings/sound/tlv320adcx140.yaml
@@ -0,0 +1,209 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+# Copyright (C) 2019 Texas Instruments Incorporated
+%YAML 1.2
+---
+$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 address 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,gpio-config:
+ description: |
+ Defines the configuration and output drive for the General Purpose
+ Input and Output pin (GPIO1). Its value is a pair, the first value is for
+ the configuration type and the second value is for the output drive
+ type. The array is defined as <GPIO1_CFG GPIO1_DRV>
+
+ configuration for the GPIO pin can be one of the following:
+ 0 - disabled
+ 1 - GPIO1 is configured as a general-purpose output (GPO)
+ 2 - (default) GPIO1 is configured as a device interrupt output (IRQ)
+ 3 - GPIO1 is configured as a secondary ASI output (SDOUT2)
+ 4 - GPIO1 is configured as a PDM clock output (PDMCLK)
+ 8 - GPIO1 is configured as an input to control when MICBIAS turns on or
+ off (MICBIAS_EN)
+ 9 - GPIO1 is configured as a general-purpose input (GPI)
+ 10 - GPIO1 is configured as a master clock input (MCLK)
+ 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN)
+ 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2
+ (PDMDIN1)
+ 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4
+ (PDMDIN2)
+ 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6
+ (PDMDIN3)
+ 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8
+ (PDMDIN4)
+
+ output drive type for the GPIO pin can be one of the following:
+ 0 - Hi-Z output
+ 1 - Drive active low and active high
+ 2 - (default) Drive active low and weak high
+ 3 - Drive active low and Hi-Z
+ 4 - Drive weak low and active high
+ 5 - Drive Hi-Z and active high
+
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 2
+ maxItems: 2
+ items:
+ maximum: 15
+ default: [2, 2]
+
+ ti,asi-tx-drive:
+ type: boolean
+ description: |
+ When set the device will set the Tx ASI output to a Hi-Z state for unused
+ data cycles. Default is to drive the output low on unused ASI cycles.
+
+patternProperties:
+ '^ti,gpo-config-[1-4]$':
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description: |
+ Defines the configuration and output driver for the general purpose
+ output pins (GPO). These values are pairs, the first value is for the
+ configuration type and the second value is for the output drive type.
+ The array is defined as <GPO_CFG GPO_DRV>
+
+ GPO output configuration can be one of the following:
+
+ 0 - (default) disabled
+ 1 - GPOX is configured as a general-purpose output (GPO)
+ 2 - GPOX is configured as a device interrupt output (IRQ)
+ 3 - GPOX is configured as a secondary ASI output (SDOUT2)
+ 4 - GPOX is configured as a PDM clock output (PDMCLK)
+
+ GPO output drive configuration for the GPO pins can be one of the following:
+
+ 0d - (default) Hi-Z output
+ 1d - Drive active low and active high
+ 2d - Drive active low and weak high
+ 3d - Drive active low and Hi-Z
+ 4d - Drive weak low and active high
+ 5d - Drive Hi-Z and active high
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ #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/Bindings/sound/tlv320aic31xx.txt b/Bindings/sound/tlv320aic31xx.txt
new file mode 100644
index 0000000..bbad98d
--- /dev/null
+++ b/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/Bindings/sound/tpa6130a2.txt b/Bindings/sound/tpa6130a2.txt
new file mode 100644
index 0000000..6dfa740
--- /dev/null
+++ b/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/Bindings/sound/tscs42xx.txt b/Bindings/sound/tscs42xx.txt
new file mode 100644
index 0000000..7eea32e
--- /dev/null
+++ b/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/Bindings/sound/tscs454.txt b/Bindings/sound/tscs454.txt
new file mode 100644
index 0000000..3ba3e2d
--- /dev/null
+++ b/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/Bindings/sound/uda1334.txt b/Bindings/sound/uda1334.txt
new file mode 100644
index 0000000..f64071b
--- /dev/null
+++ b/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/Bindings/sound/ux500-mop500.txt b/Bindings/sound/ux500-mop500.txt
new file mode 100644
index 0000000..48e071c
--- /dev/null
+++ b/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/Bindings/sound/ux500-msp.txt b/Bindings/sound/ux500-msp.txt
new file mode 100644
index 0000000..7dd1b96
--- /dev/null
+++ b/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/Bindings/sound/widgets.txt b/Bindings/sound/widgets.txt
new file mode 100644
index 0000000..b6de5ba
--- /dev/null
+++ b/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/Bindings/sound/wlf,arizona.yaml b/Bindings/sound/wlf,arizona.yaml
new file mode 100644
index 0000000..8156f30
--- /dev/null
+++ b/Bindings/sound/wlf,arizona.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/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
+
+allOf:
+ - $ref: dai-common.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.
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 1
+ maxItems: 12
+
+additionalProperties: true
diff --git a/Bindings/sound/wlf,wm8510.yaml b/Bindings/sound/wlf,wm8510.yaml
new file mode 100644
index 0000000..6d12b0a
--- /dev/null
+++ b/Bindings/sound/wlf,wm8510.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8510.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8510 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8510
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "wlf,wm8510";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8523.yaml b/Bindings/sound/wlf,wm8523.yaml
new file mode 100644
index 0000000..decc395
--- /dev/null
+++ b/Bindings/sound/wlf,wm8523.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8523.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8523 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8523
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "wlf,wm8523";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8524.yaml b/Bindings/sound/wlf,wm8524.yaml
new file mode 100644
index 0000000..4d951ec
--- /dev/null
+++ b/Bindings/sound/wlf,wm8524.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8524.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8524 24-bit 192KHz Stereo DAC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8524
+
+ "#sound-dai-cells":
+ const: 0
+
+ wlf,mute-gpios:
+ maxItems: 1
+ description:
+ a GPIO spec for the MUTE pin.
+
+required:
+ - compatible
+ - wlf,mute-gpios
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ wm8524: codec {
+ compatible = "wlf,wm8524";
+ wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>;
+ };
diff --git a/Bindings/sound/wlf,wm8580.yaml b/Bindings/sound/wlf,wm8580.yaml
new file mode 100644
index 0000000..2f27852
--- /dev/null
+++ b/Bindings/sound/wlf,wm8580.yaml
@@ -0,0 +1,42 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8580.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8580 and WM8581 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - wlf,wm8580
+ - wlf,wm8581
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "wlf,wm8580";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8711.yaml b/Bindings/sound/wlf,wm8711.yaml
new file mode 100644
index 0000000..ecaac28
--- /dev/null
+++ b/Bindings/sound/wlf,wm8711.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8711.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8711 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8711
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "wlf,wm8711";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8728.yaml b/Bindings/sound/wlf,wm8728.yaml
new file mode 100644
index 0000000..fc89475
--- /dev/null
+++ b/Bindings/sound/wlf,wm8728.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8728.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8728 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8728
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "wlf,wm8728";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8731.yaml b/Bindings/sound/wlf,wm8731.yaml
new file mode 100644
index 0000000..858c0f6
--- /dev/null
+++ b/Bindings/sound/wlf,wm8731.yaml
@@ -0,0 +1,99 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8731.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson Microelectromics WM8731 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+description: |
+ Wolfson Microelectronics WM8731 audio CODEC
+
+ Pins on the device (for linking into audio routes):
+ * LOUT: Left Channel Line Output
+ * ROUT: Right Channel Line Output
+ * LHPOUT: Left Channel Headphone Output
+ * RHPOUT: Right Channel Headphone Output
+ * LLINEIN: Left Channel Line Input
+ * RLINEIN: Right Channel Line Input
+ * MICIN: Microphone Input
+
+properties:
+ compatible:
+ enum:
+ - wlf,wm8731
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ description: Clock provider for MCLK pin.
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: mclk
+
+ AVDD-supply:
+ description: Analog power supply regulator on the AVDD pin.
+
+ HPVDD-supply:
+ description: Headphone power supply regulator on the HPVDD pin.
+
+ DBVDD-supply:
+ description: Digital buffer supply regulator for the DBVDD pin.
+
+ DCVDD-supply:
+ description: Digital core supply regulator for the DCVDD pin.
+
+required:
+ - reg
+ - compatible
+ - AVDD-supply
+ - HPVDD-supply
+ - DBVDD-supply
+ - DCVDD-supply
+
+allOf:
+ - $ref: dai-common.yaml#
+ - $ref: /schemas/spi/spi-peripheral-props.yaml#
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ wm8731_i2c: codec@0 {
+ compatible = "wlf,wm8731";
+ reg = <0>;
+ spi-max-frequency = <12500000>;
+
+ AVDD-supply = <&avdd_reg>;
+ HPVDD-supply = <&hpvdd_reg>;
+ DCVDD-supply = <&dcvdd_reg>;
+ DBVDD-supply = <&dbvdd_reg>;
+ };
+ };
+ - |
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ wm8731_spi: codec@1b {
+ compatible = "wlf,wm8731";
+ reg = <0x1b>;
+
+ AVDD-supply = <&avdd_reg>;
+ HPVDD-supply = <&hpvdd_reg>;
+ DCVDD-supply = <&dcvdd_reg>;
+ DBVDD-supply = <&dbvdd_reg>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8737.yaml b/Bindings/sound/wlf,wm8737.yaml
new file mode 100644
index 0000000..12d8765
--- /dev/null
+++ b/Bindings/sound/wlf,wm8737.yaml
@@ -0,0 +1,40 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8737.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8737 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8737
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "wlf,wm8737";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8753.yaml b/Bindings/sound/wlf,wm8753.yaml
new file mode 100644
index 0000000..9eebe7d
--- /dev/null
+++ b/Bindings/sound/wlf,wm8753.yaml
@@ -0,0 +1,62 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8753.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8753 audio CODEC
+
+description: |
+ Pins on the device (for linking into audio routes):
+ * LOUT1
+ * LOUT2
+ * ROUT1
+ * ROUT2
+ * MONO1
+ * MONO2
+ * OUT3
+ * OUT4
+ * LINE1
+ * LINE2
+ * RXP
+ * RXN
+ * ACIN
+ * ACOP
+ * MIC1N
+ * MIC1
+ * MIC2N
+ * MIC2
+ * Mic Bias
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8753
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@1a {
+ compatible = "wlf,wm8753";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8903.yaml b/Bindings/sound/wlf,wm8903.yaml
new file mode 100644
index 0000000..4cfa66f
--- /dev/null
+++ b/Bindings/sound/wlf,wm8903.yaml
@@ -0,0 +1,116 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ description: |
+ minItems: 5
+ maxItems: 5
+ A list of GPIO configuration register values.
+ If absent, no configuration of these registers is performed.
+ If any entry has the value 0xffffffff, that GPIO's
+ configuration will not be modified.
+
+ AVDD-supply:
+ description: Analog power supply regulator on the AVDD pin.
+
+ CPVDD-supply:
+ description: Charge pump supply regulator on the CPVDD pin.
+
+ DBVDD-supply:
+ description: Digital buffer supply regulator for the DBVDD pin.
+
+ DCVDD-supply:
+ description: Digital core supply regulator for the DCVDD pin.
+
+
+required:
+ - compatible
+ - reg
+ - gpio-controller
+ - '#gpio-cells'
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ wm8903: codec@1a {
+ compatible = "wlf,wm8903";
+ reg = <0x1a>;
+ interrupts = <347>;
+
+ AVDD-supply = <&fooreg_a>;
+ CPVDD-supply = <&fooreg_b>;
+ DBVDD-supply = <&fooreg_c>;
+ DCVDD-supply = <&fooreg_d>;
+
+ gpio-controller;
+ #gpio-cells = <2>;
+
+ micdet-cfg = <0>;
+ micdet-delay = <100>;
+ gpio-cfg = <
+ 0x0600 /* DMIC_LR, output */
+ 0x0680 /* DMIC_DAT, input */
+ 0x0000 /* GPIO, output, low */
+ 0x0200 /* Interrupt, output */
+ 0x01a0 /* BCLK, input, active high */
+ >;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8904.yaml b/Bindings/sound/wlf,wm8904.yaml
new file mode 100644
index 0000000..329260c
--- /dev/null
+++ b/Bindings/sound/wlf,wm8904.yaml
@@ -0,0 +1,74 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8904.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8904/WM8912 audio codecs
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+description: |
+ Pins on the device (for linking into audio routes):
+ IN1L, IN1R, IN2L, IN2R, IN3L, IN3R, HPOUTL, HPOUTR, LINEOUTL, LINEOUTR,
+ MICBIAS
+
+properties:
+ compatible:
+ enum:
+ - wlf,wm8904
+ - wlf,wm8912
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: mclk
+
+ AVDD-supply: true
+ CPVDD-supply: true
+ DBVDD-supply: true
+ DCVDD-supply: true
+ MICVDD-supply: true
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - AVDD-supply
+ - CPVDD-supply
+ - DBVDD-supply
+ - DCVDD-supply
+ - MICVDD-supply
+
+allOf:
+ - $ref: dai-common.yaml#
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "wlf,wm8904";
+ reg = <0x1a>;
+ clocks = <&pck0>;
+ clock-names = "mclk";
+ AVDD-supply = <®_1p8v>;
+ CPVDD-supply = <®_1p8v>;
+ DBVDD-supply = <®_1p8v>;
+ DCVDD-supply = <®_1p8v>;
+ MICVDD-supply = <®_1p8v>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8940.yaml b/Bindings/sound/wlf,wm8940.yaml
new file mode 100644
index 0000000..3e80921
--- /dev/null
+++ b/Bindings/sound/wlf,wm8940.yaml
@@ -0,0 +1,60 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8940.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8940 Codec
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ '#sound-dai-cells':
+ const: 0
+
+ compatible:
+ const: wlf,wm8940
+
+ reg:
+ maxItems: 1
+
+ spi-max-frequency:
+ maximum: 526000
+
+required:
+ - '#sound-dai-cells'
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@0 {
+ #sound-dai-cells = <0>;
+ compatible = "wlf,wm8940";
+ reg = <0>;
+ spi-max-frequency = <500000>;
+ };
+ };
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ #sound-dai-cells = <0>;
+ compatible = "wlf,wm8940";
+ reg = <0x1a>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/wlf,wm8960.yaml b/Bindings/sound/wlf,wm8960.yaml
new file mode 100644
index 0000000..62e62c3
--- /dev/null
+++ b/Bindings/sound/wlf,wm8960.yaml
@@ -0,0 +1,108 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8960.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8960 audio codec
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+properties:
+ compatible:
+ const: wlf,wm8960
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: mclk
+
+ '#sound-dai-cells':
+ const: 0
+
+ AVDD-supply:
+ description: Analogue supply.
+
+ DBVDD-supply:
+ description: Digital Buffer Supply.
+
+ DCVDD-supply:
+ description: Digital Core Supply.
+
+ SPKVDD1-supply:
+ description: Supply for speaker drivers 1.
+
+ SPKVDD2-supply:
+ description: Supply for speaker drivers 2.
+
+ wlf,capless:
+ type: boolean
+ description:
+ If present, OUT3 pin will be enabled and disabled together with HP_L and
+ HP_R pins in response to jack detect events.
+
+ wlf,gpio-cfg:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ maxItems: 2
+ description: |
+ A list of GPIO configuration register values.
+ - gpio-cfg[0]: ALRCGPIO of R9 (Audio interface)
+ - gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
+
+ wlf,hp-cfg:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ maxItems: 3
+ description: |
+ A list of headphone jack detect configuration register values:
+ - hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4).
+ - hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
+ - hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
+
+ wlf,shared-lrclk:
+ type: boolean
+ description:
+ If present, the LRCM bit of R24 (Additional control 2) gets set,
+ indicating that ADCLRC and DACLRC pins will be disabled only when ADC
+ (Left and Right) and DAC (Left and Right) are disabled.
+ When WM8960 works on synchronize mode and DACLRC pin is used to supply
+ frame clock, it will no frame clock for captrue unless enable DAC to
+ enable DACLRC pin. If shared-lrclk is present, no need to enable DAC for
+ captrue.
+
+required:
+ - compatible
+ - reg
+
+allOf:
+ - $ref: dai-common.yaml#
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@1a {
+ compatible = "wlf,wm8960";
+ reg = <0x1a>;
+ clocks = <&clks 0>;
+ clock-names = "mclk";
+ #sound-dai-cells = <0>;
+ wlf,hp-cfg = <3 2 3>;
+ wlf,gpio-cfg = <1 3>;
+ wlf,shared-lrclk;
+ DCVDD-supply = <®_audio>;
+ DBVDD-supply = <®_audio>;
+ AVDD-supply = <®_audio>;
+ SPKVDD1-supply = <®_audio>;
+ SPKVDD2-supply = <®_audio>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8961.yaml b/Bindings/sound/wlf,wm8961.yaml
new file mode 100644
index 0000000..f580785
--- /dev/null
+++ b/Bindings/sound/wlf,wm8961.yaml
@@ -0,0 +1,43 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8961.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8961 Ultra-Low Power Stereo CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8961
+
+ reg:
+ maxItems: 1
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+ - '#sound-dai-cells'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ wm8961: codec@4a {
+ compatible = "wlf,wm8961";
+ reg = <0x4a>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8962.yaml b/Bindings/sound/wlf,wm8962.yaml
new file mode 100644
index 0000000..5fe0b2c
--- /dev/null
+++ b/Bindings/sound/wlf,wm8962.yaml
@@ -0,0 +1,124 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8962.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8962 Ultra-Low Power Stereo CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8962
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ AVDD-supply:
+ description: Analogue supply.
+
+ CPVDD-supply:
+ description: Charge pump power supply.
+
+ DBVDD-supply:
+ description: Digital Buffer Supply.
+
+ DCVDD-supply:
+ description: Digital Core Supply.
+
+ MICVDD-supply:
+ description: Microphone bias amp supply.
+
+ PLLVDD-supply:
+ description: PLL Supply
+
+ SPKVDD1-supply:
+ description: Supply for left speaker drivers.
+
+ SPKVDD2-supply:
+ description: Supply for right speaker drivers.
+
+ spk-mono:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description:
+ If present, the SPK_MONO bit of R51 (Class D Control 2) gets set,
+ indicating that the speaker is in mono mode.
+
+ mic-cfg:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ Default register value for R48 (Additional Control 4).
+ If absent, the default should be the register default.
+
+ gpio-cfg:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ minItems: 6
+ maxItems: 6
+ description:
+ A list of GPIO configuration register values. If absent, no
+ configuration of these registers is performed. Note that only values
+ within [0x0, 0xffff] are valid. Any other value is regarded as setting
+ the GPIO register to its reset value 0x0.
+
+ port:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+ - AVDD-supply
+ - CPVDD-supply
+ - DBVDD-supply
+ - DCVDD-supply
+ - MICVDD-supply
+ - PLLVDD-supply
+ - SPKVDD1-supply
+ - SPKVDD2-supply
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/imx6qdl-clock.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ wm8962: codec@1a {
+ compatible = "wlf,wm8962";
+ reg = <0x1a>;
+ clocks = <&clks IMX6QDL_CLK_CKO>;
+ DCVDD-supply = <®_audio>;
+ DBVDD-supply = <®_audio>;
+ AVDD-supply = <®_audio>;
+ CPVDD-supply = <®_audio>;
+ MICVDD-supply = <®_audio>;
+ PLLVDD-supply = <®_audio>;
+ SPKVDD1-supply = <®_audio>;
+ SPKVDD2-supply = <®_audio>;
+ gpio-cfg = <
+ 0x0000 /* 0:Default */
+ 0x0000 /* 1:Default */
+ 0x0013 /* 2:FN_DMICCLK */
+ 0x0000 /* 3:Default */
+ 0x8014 /* 4:FN_DMICCDAT */
+ 0x0000 /* 5:Default */
+ >;
+ };
+ };
diff --git a/Bindings/sound/wlf,wm8974.txt b/Bindings/sound/wlf,wm8974.txt
new file mode 100644
index 0000000..01d3a7c
--- /dev/null
+++ b/Bindings/sound/wlf,wm8974.txt
@@ -0,0 +1,15 @@
+WM8974 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+ - compatible: "wlf,wm8974"
+ - reg: the I2C address or SPI chip select number of the device
+
+Examples:
+
+codec: wm8974@1a {
+ compatible = "wlf,wm8974";
+ reg = <0x1a>;
+};
diff --git a/Bindings/sound/wlf,wm8978.yaml b/Bindings/sound/wlf,wm8978.yaml
new file mode 100644
index 0000000..efb5f9f
--- /dev/null
+++ b/Bindings/sound/wlf,wm8978.yaml
@@ -0,0 +1,61 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM8978 Codec
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ '#sound-dai-cells':
+ const: 0
+
+ compatible:
+ const: wlf,wm8978
+
+ reg:
+ maxItems: 1
+
+ spi-max-frequency:
+ maximum: 526000
+
+required:
+ - '#sound-dai-cells'
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ spi {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@0 {
+ #sound-dai-cells = <0>;
+ compatible = "wlf,wm8978";
+ reg = <0>;
+ spi-max-frequency = <500000>;
+ };
+ };
+
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ #sound-dai-cells = <0>;
+ compatible = "wlf,wm8978";
+ reg = <0x1a>;
+ };
+ };
+
+...
diff --git a/Bindings/sound/wlf,wm8994.yaml b/Bindings/sound/wlf,wm8994.yaml
new file mode 100644
index 0000000..8f045de
--- /dev/null
+++ b/Bindings/sound/wlf,wm8994.yaml
@@ -0,0 +1,194 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8994.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Wolfson WM1811/WM8994/WM8958 audio codecs
+
+maintainers:
+ - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+ - patches@opensource.cirrus.com
+
+description: |
+ These devices support both I2C and SPI (configured with pin strapping on the
+ board).
+
+ Pins on the device (for linking into audio routes):
+ IN1LN, IN1LP, IN2LN, IN2LP:VXRN, IN1RN, IN1RP, IN2RN, IN2RP:VXRP, SPKOUTLP,
+ SPKOUTLN, SPKOUTRP, SPKOUTRN, HPOUT1L, HPOUT1R, HPOUT2P, HPOUT2N, LINEOUT1P,
+ LINEOUT1N, LINEOUT2P, LINEOUT2N.
+
+properties:
+ compatible:
+ enum:
+ - wlf,wm1811
+ - wlf,wm8994
+ - wlf,wm8958
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ minItems: 1
+ maxItems: 2
+
+ clock-names:
+ minItems: 1
+ items:
+ - const: MCLK1
+ - const: MCLK2
+
+ gpio-controller: true
+
+ '#gpio-cells':
+ const: 2
+
+ interrupts:
+ maxItems: 1
+
+ interrupt-controller: true
+
+ '#interrupt-cells':
+ const: 2
+ description:
+ The first cell is the IRQ number. The second cell is the flags, encoded
+ as the trigger masks.
+
+ AVDD1-supply: true
+ AVDD2-supply: true
+ CPVDD-supply: true
+ DBVDD-supply: true
+ DBVDD1-supply: true
+ DBVDD2-supply: true
+ DBVDD3-supply: true
+ DCVDD-supply: true
+ LDO1VDD-supply: true
+ LDO2VDD-supply: true
+ SPKVDD1-supply: true
+ SPKVDD2-supply: true
+
+ '#sound-dai-cells':
+ const: 0
+
+ wlf,gpio-cfg:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ maxItems: 11
+ description:
+ A list of GPIO configuration register values. If absent, no configuration
+ of these registers is performed. If any value is over 0xffff then the
+ register will be left as default. If present 11 values must be supplied.
+
+ wlf,micbias-cfg:
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ maxItems: 2
+ description:
+ Two MICBIAS register values for WM1811 or WM8958. If absent the register
+ defaults will be used.
+
+ wlf,ldo1ena-gpios:
+ maxItems: 1
+ description:
+ Control of LDO1ENA input to device.
+
+ wlf,ldo2ena-gpios:
+ maxItems: 1
+ description:
+ Control of LDO2ENA input to device.
+
+ wlf,lineout1-se:
+ type: boolean
+ description:
+ LINEOUT1 is in single ended mode.
+
+ wlf,lineout2-se:
+ type: boolean
+ description:
+ INEOUT2 is in single ended mode.
+
+ wlf,lineout1-feedback:
+ type: boolean
+ description:
+ LINEOUT1 has common mode feedback connected.
+
+ wlf,lineout2-feedback:
+ type: boolean
+ description:
+ LINEOUT2 has common mode feedback connected.
+
+ wlf,ldoena-always-driven:
+ type: boolean
+ description:
+ LDOENA is always driven.
+
+ wlf,spkmode-pu:
+ type: boolean
+ description:
+ Enable the internal pull-up resistor on the SPKMODE pin.
+
+ wlf,csnaddr-pd:
+ type: boolean
+ description:
+ Enable the internal pull-down resistor on the CS/ADDR pin.
+
+required:
+ - compatible
+ - reg
+ - AVDD2-supply
+ - CPVDD-supply
+ - SPKVDD1-supply
+ - SPKVDD2-supply
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ enum:
+ - wlf,wm1811
+ - wlf,wm8958
+ then:
+ properties:
+ DBVDD-supply: false
+ LDO2VDD-supply: false
+ required:
+ - DBVDD1-supply
+ - DBVDD2-supply
+ - DBVDD3-supply
+ else:
+ properties:
+ DBVDD1-supply: false
+ DBVDD2-supply: false
+ DBVDD3-supply: false
+ required:
+ - DBVDD-supply
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ audio-codec@1a {
+ compatible = "wlf,wm1811";
+ reg = <0x1a>;
+ clocks = <&i2s0 0>;
+ clock-names = "MCLK1";
+
+ AVDD2-supply = <&main_dc_reg>;
+ CPVDD-supply = <&main_dc_reg>;
+ DBVDD1-supply = <&main_dc_reg>;
+ DBVDD2-supply = <&main_dc_reg>;
+ DBVDD3-supply = <&main_dc_reg>;
+ LDO1VDD-supply = <&main_dc_reg>;
+ SPKVDD1-supply = <&main_dc_reg>;
+ SPKVDD2-supply = <&main_dc_reg>;
+
+ wlf,ldo1ena-gpios = <&gpb0 0 GPIO_ACTIVE_HIGH>;
+ wlf,ldo2ena-gpios = <&gpb0 1 GPIO_ACTIVE_HIGH>;
+ };
+ };
diff --git a/Bindings/sound/wm8741.txt b/Bindings/sound/wm8741.txt
new file mode 100644
index 0000000..b69e196
--- /dev/null
+++ b/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/Bindings/sound/wm8750.yaml b/Bindings/sound/wm8750.yaml
new file mode 100644
index 0000000..24246ac
--- /dev/null
+++ b/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/Bindings/sound/wm8770.txt b/Bindings/sound/wm8770.txt
new file mode 100644
index 0000000..cac762a
--- /dev/null
+++ b/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/Bindings/sound/wm8776.txt b/Bindings/sound/wm8776.txt
new file mode 100644
index 0000000..0117336
--- /dev/null
+++ b/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/Bindings/sound/wm8782.txt b/Bindings/sound/wm8782.txt
new file mode 100644
index 0000000..1a28f32
--- /dev/null
+++ b/Bindings/sound/wm8782.txt
@@ -0,0 +1,24 @@
+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)
+
+Optional properties:
+
+ - wlf,fsampen:
+ FSAMPEN pin value, 0 for low, 1 for high, 2 for disconnected.
+ Defaults to 0 if left unspecified.
+
+Example:
+
+wm8782: stereo-adc {
+ compatible = "wlf,wm8782";
+ Vdda-supply = <&vdda_supply>;
+ Vdd-supply = <&vdd_supply>;
+ wlf,fsampen = <2>; /* 192KHz */
+};
diff --git a/Bindings/sound/wm8804.txt b/Bindings/sound/wm8804.txt
new file mode 100644
index 0000000..2c1641c
--- /dev/null
+++ b/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/Bindings/sound/xlnx,audio-formatter.txt b/Bindings/sound/xlnx,audio-formatter.txt
new file mode 100644
index 0000000..cbc93c8
--- /dev/null
+++ b/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/Bindings/sound/xlnx,i2s.txt b/Bindings/sound/xlnx,i2s.txt
new file mode 100644
index 0000000..5e7c7d5
--- /dev/null
+++ b/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/Bindings/sound/xlnx,spdif.txt b/Bindings/sound/xlnx,spdif.txt
new file mode 100644
index 0000000..15c2d64
--- /dev/null
+++ b/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/Bindings/sound/zl38060.yaml b/Bindings/sound/zl38060.yaml
new file mode 100644
index 0000000..8bd201e
--- /dev/null
+++ b/Bindings/sound/zl38060.yaml
@@ -0,0 +1,72 @@
+# 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>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+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'
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ spi {
+ #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>;
+ };
+ };