diff options
| author | Niklas Cassel <cassel@kernel.org> | 2026-09-16 15:00:32 +0200 |
|---|---|---|
| committer | Niklas Cassel <cassel@kernel.org> | 2026-09-16 16:18:00 +0200 |
| commit | 8d836581f9b1f57ceaa2b47654754ef1260b410b (patch) | |
| tree | 1dbd4d0dfb99c2e238356cbe947e5b0be3e3a571 /Documentation/devicetree/bindings/sound | |
| download | linux-stable-8d836581f9b1f57ceaa2b47654754ef1260b410b.tar.gz linux-stable-8d836581f9b1f57ceaa2b47654754ef1260b410b.zip | |
ata: libata-scsi: fix ata_dsm_trim_pages() kernel-docgrafted
make htmldocs fails with:
Documentation/driver-api/libata:604: ./drivers/ata/libata-scsi.c:2301:
ERROR: Unexpected indentation. [docutils]
The Return: section of ata_dsm_trim_pages() ends its first sentence with
a colon and continues with an indented bullet list. reStructuredText
requires a blank line before an indented block, so docutils chokes on
the list.
Simply adding the missing blank line does not work either: Return: is a
kernel-doc "special section", which is terminated by the first blank
line, so the bullet list would end up in the Description section,
detached from the sentence introducing it.
Spell the two bounds out as prose instead, so that the Return: section
stays self-contained. Documentation-only change.
Fixes: e64e6b5dc867 ("ata: libata-scsi: scale DSM TRIM payload by MAX PAGES PER DSM COMMAND")
Reported-by: Thomas Huth <thuth@redhat.com>
Closes: https://lore.kernel.org/linux-ide/b0a0b8e8-cc4f-4c7b-8bc5-fee0712d405a@redhat.com/
Reviewed-by: Damien Le Moal <dlemoal@kernel.org>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Link: https://lore.kernel.org/r/20260916130031.29990-2-cassel@kernel.org
Signed-off-by: Niklas Cassel <cassel@kernel.org>
Diffstat (limited to 'Documentation/devicetree/bindings/sound')
449 files changed, 42622 insertions, 0 deletions
diff --git a/Documentation/devicetree/bindings/sound/ac97-bus.txt b/Documentation/devicetree/bindings/sound/ac97-bus.txt new file mode 100644 index 000000000..103c428f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ac97-bus.txt @@ -0,0 +1,32 @@ +Generic AC97 Device Properties + +This documents describes the devicetree bindings for an ac97 controller child +node describing ac97 codecs. + +Required properties: +-compatible : Must be "ac97,vendor_id1,vendor_id2 + The ids shall be the 4 characters hexadecimal encoding, such as + given by "%04x" formatting of printf +-reg : Must be the ac97 codec number, between 0 and 3 + +Example: +ac97: sound@40500000 { + compatible = "marvell,pxa270-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pinctrl_ac97_default >; + clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>; + clock-names = "AC97CLK", "AC97CONFCLK"; + + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + reg = <0>; + compatible = "ac97,574d,4c13"; + clocks = <&fixed_wm9713_clock>; + clock-names = "ac97_clk"; + } +}; diff --git a/Documentation/devicetree/bindings/sound/adi,adau1372.yaml b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml new file mode 100644 index 000000000..9a7ff50a0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1372.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1372 CODEC + +maintainers: + - Alexandre Belloni <alexandre.belloni@bootlin.com> + +description: | + Analog Devices ADAU1372 four inputs and two outputs codec. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,adau1372 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + powerdown-gpios: + description: GPIO used for hardware power-down. + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@3c { + compatible = "adi,adau1372"; + reg = <0x3c>; + #sound-dai-cells = <0>; + clock-names = "mclk"; + clocks = <&adau1372z_xtal>; + }; + }; + + adau1372z_xtal: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <12288000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/adi,adau1373.yaml b/Documentation/devicetree/bindings/sound/adi,adau1373.yaml new file mode 100644 index 000000000..97552bf5d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1373.yaml @@ -0,0 +1,111 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1373.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1373 CODEC + +maintainers: + - Nuno Sá <nuno.sa@analog.com> + +description: | + Analog Devices ADAU1373 Low power codec with speaker and headphone amplifiers. + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1373.pdf + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,adau1373 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + powerdown-gpios: + description: GPIO used for hardware power-down. + maxItems: 1 + + adi,micbias1-microvolt: + description: + This property sets the microphone bias voltage for the first microphone. + enum: [1800000, 2200000, 2600000, 2900000] + default: 2900000 + + adi,micbias2-microvolt: + description: + This property sets the microphone bias voltage for the second microphone. + enum: [1800000, 2200000, 2600000, 2900000] + default: 2900000 + + adi,input1-differential: + description: This property sets the first analog input as differential. + type: boolean + + adi,input2-differential: + description: This property sets the second analog input as differential. + type: boolean + + adi,input3-differential: + description: This property sets the third analog input as differential. + type: boolean + + adi,input4-differential: + description: This property sets the fourth analog input as differential. + type: boolean + + adi,lineout-differential: + description: This property sets the line output as differential. + type: boolean + + adi,lineout-gnd-sense: + description: This property enables the line output ground sense control. + type: boolean + + adi,drc-settings: + description: + This setting is used to control the dynamic range of the signal. The + device provides a maximum of three full band DRCs with 13 entries each. + $ref: /schemas/types.yaml#/definitions/uint8-array + oneOf: + - minItems: 13 + maxItems: 13 + - minItems: 26 + maxItems: 26 + - minItems: 39 + maxItems: 39 + +required: + - "#sound-dai-cells" + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@1a { + compatible = "adi,adau1373"; + reg = <0x1a>; + #sound-dai-cells = <0>; + powerdown-gpios = <&gpio 100 GPIO_ACTIVE_LOW>; + adi,input2-differential; + adi,input1-differential; + adi,lineout-differential; + adi,micbias2-microvolt = <1800000>; + adi,drc-settings = /bits/ 8 < + 0xff 0xff 0x1 0x2 0xa 0xa 0xd 0x1 0xff 0xff 0x5 0xd 0xff + >; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt new file mode 100644 index 000000000..0d1128ce2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -0,0 +1,39 @@ +Analog Devices ADAU1701 + +Required properties: + + - compatible: Should contain "adi,adau1701" + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + - adi,pll-mode-gpios: An array of two GPIO specs to describe the GPIOs + the ADAU's PLL config pins are connected to. + The state of the pins are set according to the + configured clock divider on ASoC side before the + firmware is loaded. + - adi,pin-config: An array of 12 numerical values selecting one of the + pin configurations as described in the datasheet, + table 53. Note that the value of this property has + to be prefixed with '/bits/ 8'. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + +Examples: + + i2c_bus { + adau1701@34 { + compatible = "adi,adau1701"; + reg = <0x34>; + reset-gpio = <&gpio 23 0>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; + adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>; + adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4 + 0x4 0x4 0x4 0x4 0x4 0x4>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.yaml b/Documentation/devicetree/bindings/sound/adi,adau17x1.yaml new file mode 100644 index 000000000..8ef1e7f6e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau17x1.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781 Codec + +maintainers: + - Lars-Peter Clausen <lars@metafoo.de> + +properties: + compatible: + enum: + - adi,adau1361 + - adi,adau1381 + - adi,adau1461 + - adi,adau1761 + - adi,adau1781 + - adi,adau1961 + + reg: + maxItems: 1 + description: + The i2c address. Value depends on the state of ADDR0 and ADDR1, + as wired in hardware. + + clock-names: + const: mclk + + clocks: + items: + - description: provides the audio master clock for the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@38 { + compatible = "adi,adau1761"; + reg = <0x38>; + clock-names = "mclk"; + clocks = <&audio_clock>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml new file mode 100644 index 000000000..dba3023a4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1977.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1977/ADAU1978/ADAU1979 Quad ADC with Diagnostics + +maintainers: + - Lars-Peter Clausen <lars@metafoo.de> + - Bogdan Togorean <bogdan.togorean@analog.com> + +description: | + Analog Devices ADAU1977 and similar quad ADC with Diagnostics + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf + +properties: + compatible: + enum: + - adi,adau1977 + - adi,adau1978 + - adi,adau1979 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + reset-gpios: + maxItems: 1 + + AVDD-supply: + description: Analog power support for the device. + + DVDD-supply: + description: Supply voltage for digital core. + + adi,micbias: + description: | + Configures the voltage setting for the MICBIAS pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3, 4, 5, 6, 7, 8] + default: 7 + +required: + - reg + - compatible + - AVDD-supply + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + spi { + #address-cells = <1>; + #size-cells = <0>; + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + reg = <0>; + spi-max-frequency = <600000>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + + adi,micbias = <3>; + }; + }; + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <®ulator>; + DVDD-supply = <®ulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.yaml b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml new file mode 100644 index 000000000..7858f3f8e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,adau7002 + + IOVDD-supply: + description: + IOVDD power supply, if skipped then it is assumed that the supply pin is + hardwired to always on. + + wakeup-delay-ms: + description: + Delay after power up needed for device to settle. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + audio-codec { + compatible = "adi,adau7002"; + IOVDD-supply = <&pp1800_l15a>; + #sound-dai-cells = <0>; + wakeup-delay-ms = <80>; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml new file mode 100644 index 000000000..11f59c29b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 000000000..229ad1392 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,34 @@ +ADI AXI-I2S controller + +The core can be generated with transmit (playback), only receive +(capture) or both directions enabled. + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects two dma channels if both transmit and receive are + enabled, one channel otherwise. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + i2s: i2s@77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>, <&ps7_dma 1>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt new file mode 100644 index 000000000..7b664e7cb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt @@ -0,0 +1,30 @@ +ADI AXI-SPDIF controller + +Required properties: + - compatible : Must be "adi,axi-spdif-tx-1.00.a" + - reg : Must contain SPDIF core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names: "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channel that is used by + the core. The core expects one dma channel for transmit. + - dma-names : Must be "tx" + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + spdif: spdif@77400000 { + compatible = "adi,axi-spdif-tx-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>; + dma-names = "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,max98388.yaml b/Documentation/devicetree/bindings/sound/adi,max98388.yaml new file mode 100644 index 000000000..93ccd5905 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98388.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98388.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98388 Speaker Amplifier + +maintainers: + - Ryan Lee <ryans.lee@analog.com> + +description: + The MAX98388 is a mono Class-D speaker amplifier with I/V feedback. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,max98388 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + adi,vmon-slot-no: + description: slot number of the voltage feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 1 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V feedback data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98388: amplifier@39 { + compatible = "adi,max98388"; + reg = <0x39>; + #sound-dai-cells = <0>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml new file mode 100644 index 000000000..bdc10d420 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -0,0 +1,141 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98396.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98396 Speaker Amplifier + +maintainers: + - Ryan Lee <ryans.lee@analog.com> + +description: + The MAX98396 is a mono Class-DG speaker amplifier with I/V sense. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + The MAX98397 is a variant of MAX98396 with wide input supply range. + +properties: + compatible: + enum: + - adi,max98396 + - adi,max98397 + reg: + maxItems: 1 + description: I2C address of the device. + + avdd-supply: + description: A 1.8V supply that powers up the AVDD pin. + + dvdd-supply: + description: A 1.2V supply that powers up the DVDD pin. + + dvddio-supply: + description: A 1.2V or 1.8V supply that powers up the VDDIO pin. + + pvdd-supply: + description: A 3.0V to 20V supply that powers up the PVDD pin. + + vbat-supply: + description: A 3.3V to 5.5V supply that powers up the VBAT pin. + + adi,vmon-slot-no: + description: slot number of the voltage sense monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current sense monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 1 + + adi,spkfb-slot-no: + description: slot number of speaker DSP monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 2 + + adi,bypass-slot-no: + description: + Selects the PCM data input channel that is routed to the speaker + audio processing bypass path. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 0 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V sense data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-ms: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + enum: [64, 256, 1024, 4096] + default: 64 + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98396: amplifier@39 { + compatible = "adi,max98396"; + reg = <0x39>; + dvdd-supply = <®ulator_1v2>; + dvddio-supply = <®ulator_1v8>; + avdd-supply = <®ulator_1v8>; + pvdd-supply = <®ulator_pvdd>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.yaml b/Documentation/devicetree/bindings/sound/adi,ssm2305.yaml new file mode 100644 index 000000000..b841da2dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.yaml @@ -0,0 +1,46 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,ssm2305.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices SSM2305 Class-D Speaker Amplifier + +maintainers: + - Lars-Peter Clausen <lars@metafoo.de> + +description: + The SSM2305 is a filterless, high efficiency, mono 2.8 W Class-D + audio amplifier with a micropower shutdown mode controlled via a + dedicated active-low GPIO pin. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,ssm2305 + + shutdown-gpios: + maxItems: 1 + description: + GPIO connected to the shutdown pin (SD) of the SSM2305. + The pin is active-low; asserting it puts the device into + micropower shutdown mode. + +required: + - compatible + - shutdown-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml b/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml new file mode 100644 index 000000000..f1beae84c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,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: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4118.txt b/Documentation/devicetree/bindings/sound/ak4118.txt new file mode 100644 index 000000000..6e11a2f74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4118.txt @@ -0,0 +1,22 @@ +AK4118 S/PDIF transceiver + +This device supports I2C mode. + +Required properties: + +- compatible : "asahi-kasei,ak4118" +- reg : The I2C address of the device for I2C +- reset-gpios: A GPIO specifier for the reset pin +- irq-gpios: A GPIO specifier for the IRQ pin + +Example: + +&i2c { + ak4118: ak4118@13 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4118"; + reg = <0x13>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW> + irq-gpios = <&gpio 1 GPIO_ACTIVE_HIGH>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt new file mode 100644 index 000000000..ec3df3abb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -0,0 +1,23 @@ +AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + +This device has no control interface. + +Required properties: + + - compatible : "asahi-kasei,ak5386" + +Optional properties: + + - reset-gpio : a GPIO spec for the reset/power down pin. + If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V + +Example: + +spdif: ak5386@0 { + compatible = "asahi-kasei,ak5386"; + reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000..b4eca702f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,346 @@ +# 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 + - allwinner,sun50i-h616-codec + - allwinner,suniv-f1c100s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + oneOf: + - items: + - description: RX DMA Channel + - description: TX DMA Channel + - items: + - description: TX DMA Channel + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - items: + - 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 + - MIC + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Right FM In + - Left FM In + - 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 + + hp-det-gpios: + maxItems: 1 + description: GPIO for headphone/line-out detection + +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 + + - if: + properties: + compatible: + enum: + - allwinner,sun50i-h616-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - Headphone + - LINEOUT + - Line Out + - Speaker + + dmas: + items: + - description: TX DMA Channel + + dma-names: + items: + - const: tx + + else: + properties: + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + - if: + properties: + compatible: + enum: + - allwinner,suniv-f1c100s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Right FM In + - Left FM In + - 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/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml new file mode 100644 index 000000000..ae86cb5f0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml @@ -0,0 +1,149 @@ +# 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: + - enum: + - allwinner,sun20i-d1-i2s + - allwinner,sun55i-a523-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: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-i2s + - allwinner,sun8i-a83t-i2s + - allwinner,sun8i-h3-i2s + - allwinner,sun50i-a64-codec-i2s + - allwinner,sun50i-h6-i2s + - allwinner,sun50i-r329-i2s + + then: + required: + - resets + + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun8i-a83t-i2s + - allwinner,sun8i-h3-i2s + + then: + properties: + dmas: + minItems: 1 + items: + - description: RX DMA Channel + - description: TX DMA Channel + description: + Some controllers cannot receive but can only transmit + data. In such a case, the RX DMA channel is to be omitted. + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - const: tx + description: + Some controllers cannot receive but can only transmit + data. In such a case, the RX name is to be omitted. + + else: + properties: + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + i2s0: i2s@1c22400 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-i2s"; + reg = <0x01c22400 0x400>; + interrupts = <0 16 4>; + clocks = <&apb0_gates 3>, <&i2s0_clk>; + clock-names = "apb", "mod"; + dmas = <&dma 0 3>, <&dma 0 3>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml new file mode 100644 index 000000000..1d089ba70 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -0,0 +1,158 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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 + - const: allwinner,sun50i-h616-spdif + - const: allwinner,sun55i-a523-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: + minItems: 2 + maxItems: 3 + + clock-names: + minItems: 2 + maxItems: 3 + + # 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: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-spdif + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif + - allwinner,sun50i-h616-spdif + - allwinner,sun55i-a523-spdif + + then: + required: + - resets + + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h616-spdif + + then: + properties: + dmas: + description: TX DMA Channel + + dma-names: + const: tx + + else: + properties: + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun55i-a523-spdif + + then: + properties: + clocks: + items: + - description: Bus Clock + - description: TX Clock + - description: RX Clock + + clock-names: + items: + - const: apb + - const: tx + - const: rx + else: + properties: + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: spdif + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + spdif: spdif@1c21000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-spdif"; + reg = <0x01c21000 0x40>; + interrupts = <13>; + clocks = <&apb0_gates 1>, <&spdif_clk>; + clock-names = "apb", "spdif"; + dmas = <&dma 0 2>, <&dma 0 2>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml new file mode 100644 index 000000000..5800de63f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-a64-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A64 Analog Codec + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + compatible: + const: allwinner,sun50i-a64-codec-analog + + reg: + maxItems: 1 + + cpvdd-supply: + description: + Regulator for the headphone amplifier + + allwinner,internal-bias-resistor: + description: + Enable the internal 2.2K bias resistor between HBIAS and MICDET pins + type: boolean + +required: + - compatible + - reg + - cpvdd-supply + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun50i-a64-codec-analog"; + reg = <0x01f015c0 0x4>; + cpvdd-supply = <®_eldo1>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000..763b87604 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC + +maintainers: + - Ban Tao <fengzheng923@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - items: + - enum: + - allwinner,sun20i-d1-dmic + - const: allwinner,sun50i-h6-dmic + - const: allwinner,sun50i-h6-dmic + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml new file mode 100644 index 000000000..1c21a1b39 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a23-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A23 Analog Codec + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <mripard@kernel.org> + +properties: + compatible: + oneOf: + # FIXME: This is documented in the PRCM binding, but needs to be + # migrated here at some point + # - allwinner,sun8i-a23-codec-analog + - const: allwinner,sun8i-h3-codec-analog + - items: + - const: allwinner,sun8i-v3-codec-analog + - const: allwinner,sun8i-h3-codec-analog + - const: allwinner,sun8i-v3s-codec-analog + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun8i-h3-codec-analog"; + reg = <0x01f015c0 0x4>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml new file mode 100644 index 000000000..63eadc420 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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: + - $ref: dai-common.yaml# + +properties: + "#sound-dai-cells": + minimum: 0 + maximum: 1 + description: + A value of 0 is deprecated. When used, it only allows access to + the ADC/DAC and AIF1 (the CPU DAI), not the other two AIFs/DAIs. + + compatible: + oneOf: + - items: + - const: allwinner,sun50i-a64-codec + - const: allwinner,sun8i-a33-codec + - const: allwinner,sun8i-a33-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + +unevaluatedProperties: false + +examples: + - | + audio-codec@1c22e00 { + #sound-dai-cells = <1>; + compatible = "allwinner,sun8i-a33-codec"; + reg = <0x01c22e00 0x400>; + interrupts = <0 29 4>; + clocks = <&ccu 47>, <&ccu 92>; + clock-names = "bus", "mod"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml new file mode 100644 index 000000000..6350dfc0a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -0,0 +1,118 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,aiu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AIU audio output controller + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: 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/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml new file mode 100644 index 000000000..b1b48d683 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml @@ -0,0 +1,112 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-fifo.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AXG Audio FIFO controllers + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - enum: + - amlogic,axg-toddr + - amlogic,axg-frddr + - items: + - enum: + - amlogic,g12a-toddr + - amlogic,sm1-toddr + - const: amlogic,axg-toddr + - items: + - enum: + - amlogic,g12a-frddr + - amlogic,sm1-frddr + - const: amlogic,axg-frddr + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + + interrupts: + maxItems: 1 + + resets: + minItems: 1 + maxItems: 2 + + reset-names: + minItems: 1 + maxItems: 2 + + amlogic,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Size of the controller's fifo in bytes + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - interrupts + - resets + - amlogic,fifo-depth + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-toddr + - amlogic,sm1-toddr + - amlogic,g12a-frddr + - amlogic,sm1-frddr + + then: + properties: + resets: + minItems: 2 + reset-names: + items: + - const: arb + - const: rst + required: + - reset-names + + else: + properties: + resets: + maxItems: 1 + reset-names: + const: arb + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/reset/amlogic,meson-axg-audio-arb.h> + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + audio-controller@1c0 { + compatible = "amlogic,g12a-frddr", "amlogic,axg-frddr"; + reg = <0x1c0 0x1c>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; + interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; + resets = <&arb>, <&clkc_audio AUD_RESET_FRDDR_A>; + reset-names = "arb", "rst"; + amlogic,fifo-depth = <512>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml new file mode 100644 index 000000000..df21dd72f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-pdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG PDM input + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - items: + - enum: + - amlogic,g12a-pdm + - amlogic,sm1-pdm + - const: amlogic,axg-pdm + - const: amlogic,axg-pdm + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: PDM digital clock + - description: DSP system clock + + clock-names: + items: + - const: pclk + - const: dclk + - const: sysclk + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-pdm + - amlogic,sm1-pdm + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller@ff632000 { + compatible = "amlogic,axg-pdm"; + reg = <0xff632000 0x34>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_PDM>, + <&clkc_audio AUD_CLKID_PDM_DCLK>, + <&clkc_audio AUD_CLKID_PDM_SYSCLK>; + clock-names = "pclk", "dclk", "sysclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml new file mode 100644 index 000000000..177db900d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -0,0 +1,181 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + + clocks: + minItems: 1 + maxItems: 3 + description: + Base PLL clocks of audio subsystem, used to configure base clock + frequencies for different audio use-cases. + +patternProperties: + "^dai-link-[0-9]+$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and the CODEC sub-nodes. + There should be at least one (and probably more) subnode of this type + + properties: + dai-format: + $ref: /schemas/types.yaml#/definitions/string + enum: [ i2s, left-j, dsp_a ] + + dai-tdm-slot-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of slots in use. If omitted, slot number is set to + accommodate the largest mask provided. + maximum: 32 + + dai-tdm-slot-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Width in bits for each slot + enum: [ 8, 16, 20, 24, 32 ] + default: 32 + + mclk-fs: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Multiplication factor between the frame rate and master clock + rate + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + patternProperties: + "^dai-tdm-slot-(t|r)x-mask-[0-3]$": + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 32 + description: + Transmit and receive cpu slot masks of each TDM lane + When omitted, mask is assumed to have to no slots. A valid + interface must have at least one slot, so at least one these + mask should be provided with an enabled slot. + + "^codec(-[0-9]+)?$": + type: object + additionalProperties: false + description: + dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. dai-link representing + frontend links have no codec, therefore have no subnodes + + properties: + sound-dai: + maxItems: 1 + description: phandle of the codec DAI + + patternProperties: + "^dai-tdm-slot-(t|r)x-mask$": + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 32 + description: Transmit and receive codec slot masks + + required: + - sound-dai + + required: + - sound-dai + +required: + - dai-link-0 + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "amlogic,axg-sound-card"; + model = "AXG-S420"; + audio-aux-devs = <&tdmin_a>, <&tdmout_c>; + audio-widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right", + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; + + dai-link-0 { + sound-dai = <&frddr_a>; + }; + + dai-link-1 { + sound-dai = <&toddr_a>; + }; + + dai-link-2 { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-2 = <1 1>; + dai-tdm-slot-tx-mask-3 = <1 1>; + dai-tdm-slot-rx-mask-1 = <1 1>; + mclk-fs = <256>; + + codec-0 { + sound-dai = <&lineout>; + }; + + codec-1 { + sound-dai = <&speaker_amp1>; + }; + + codec-2 { + sound-dai = <&speaker_amp2>; + }; + + codec-3 { + sound-dai = <&linein>; + }; + }; + + dai-link-3 { + sound-dai = <&spdifout>; + + codec { + sound-dai = <&spdif_dit>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml new file mode 100644 index 000000000..a0bd7a5fb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifin.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG SPDIF Input + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - const: amlogic,axg-spdifin + - items: + - enum: + - amlogic,g12a-spdifin + - amlogic,sm1-spdifin + - const: amlogic,axg-spdifin + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: SPDIF input reference clock + + clock-names: + items: + - const: pclk + - const: refclk + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + - interrupts + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-spdifin + - amlogic,sm1-spdifin + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + audio-controller@400 { + compatible = "amlogic,axg-spdifin"; + reg = <0x400 0x30>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>; + clocks = <&clkc_audio AUD_CLKID_SPDIFIN>, + <&clkc_audio AUD_CLKID_SPDIFIN_CLK>; + clock-names = "pclk", "refclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml new file mode 100644 index 000000000..15be8dae9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifout.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG SPDIF Output + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - const: amlogic,axg-spdifout + - items: + - enum: + - amlogic,g12a-spdifout + - amlogic,sm1-spdifout + - const: amlogic,axg-spdifout + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: SPDIF output master clock + + clock-names: + items: + - const: pclk + - const: mclk + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-spdifout + - amlogic,sm1-spdifout + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller@480 { + compatible = "amlogic,axg-spdifout"; + reg = <0x480 0x50>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, + <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; + clock-names = "pclk", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml new file mode 100644 index 000000000..719ca8fc9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-formatters.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG TDM formatters + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + enum: + - amlogic,g12a-tdmout + - amlogic,sm1-tdmout + - amlogic,axg-tdmout + - amlogic,g12a-tdmin + - amlogic,sm1-tdmin + - amlogic,axg-tdmin + + clocks: + items: + - description: Peripheral clock + - description: Bit clock + - description: Bit clock input multiplexer + - description: Sample clock + - description: Sample clock input multiplexer + + clock-names: + items: + - const: pclk + - const: sclk + - const: sclk_sel + - const: lrclk + - const: lrclk_sel + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + +allOf: + - $ref: component-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-tdmin + - amlogic,sm1-tdmin + - amlogic,g12a-tdmout + - amlogic,sm1-tdmout + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + audio-controller@500 { + compatible = "amlogic,g12a-tdmout"; + reg = <0x500 0x40>; + resets = <&clkc_audio AUD_RESET_TDMOUT_A>; + clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; + clock-names = "pclk", "sclk", "sclk_sel", + "lrclk", "lrclk_sel"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml new file mode 100644 index 000000000..45955d8a2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-iface.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio TDM Interfaces + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: amlogic,axg-tdm-iface + + "#sound-dai-cells": + const: 0 + + clocks: + minItems: 2 + items: + - description: Bit clock + - description: Sample clock + - description: Master clock # optional + + clock-names: + minItems: 2 + items: + - const: sclk + - const: lrclk + - const: mclk + +required: + - compatible + - "#sound-dai-cells" + - clocks + - clock-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller { + compatible = "amlogic,axg-tdm-iface"; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_MST_A_SCLK>, + <&clkc_audio AUD_CLKID_MST_A_LRCLK>, + <&clkc_audio AUD_CLKID_MST_A_MCLK>; + clock-names = "sclk", "lrclk", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml new file mode 100644 index 000000000..23f82bb89 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a Internal DAC Control Glue + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: amlogic,g12a-toacodec + - items: + - enum: + - amlogic,sm1-toacodec + - const: amlogic,g12a-toacodec + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + toacodec: audio-controller@740 { + compatible = "amlogic,g12a-toacodec"; + reg = <0x740 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOACODEC>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml new file mode 100644 index 000000000..b4b78475c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-tohdmitx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a HDMI TX Control Glue + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + compatible: + oneOf: + - items: + - const: amlogic,g12a-tohdmitx + - items: + - enum: + - amlogic,sm1-tohdmitx + - const: amlogic,g12a-tohdmitx + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - resets + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + tohdmitx: audio-controller@744 { + compatible = "amlogic,g12a-tohdmitx"; + reg = <0x744 0x4>; + resets = <&clkc_audio AUD_RESET_TOHDMITX>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml new file mode 100644 index 000000000..6fdf605ad --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -0,0 +1,113 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: |- + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + + clocks: + minItems: 1 + maxItems: 3 + description: + Base PLL clocks of audio subsystem, used to configure base clock + frequencies for different audio use-cases. + +patternProperties: + "^dai-link-[0-9]+$": + 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/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml new file mode 100644 index 000000000..5f5cccdbe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic T9015 Internal Audio DAC + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 0 + + compatible: + items: + - const: amlogic,t9015 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: pclk + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + AVDD-supply: + description: + Analogue power supply. + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - resets + - AVDD-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/g12a-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-reset.h> + + acodec: audio-controller@32000 { + compatible = "amlogic,t9015"; + reg = <0x32000 0x14>; + #sound-dai-cells = <0>; + clocks = <&clkc CLKID_AUDIO_CODEC>; + clock-names = "pclk"; + resets = <&reset RESET_AUDIO_CODEC>; + AVDD-supply = <&vddao_1v8>; + }; diff --git a/Documentation/devicetree/bindings/sound/apple,mca.yaml b/Documentation/devicetree/bindings/sound/apple,mca.yaml new file mode 100644 index 000000000..2beb72511 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/apple,mca.yaml @@ -0,0 +1,140 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/apple,mca.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Apple MCA I2S transceiver + +description: | + MCA is an I2S transceiver peripheral found on M1 and other Apple chips. It is + composed of a number of identical clusters which can operate independently + or in an interlinked fashion. Up to 6 clusters have been seen on an MCA. + +maintainers: + - Martin PoviÅ¡er <povik+lin@cutebit.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - items: + - const: apple,t6020-mca + - const: apple,t8103-mca + - items: + - enum: + # Do not add additional SoC to this list. + - apple,t6000-mca + - apple,t8103-mca + - apple,t8112-mca + - const: apple,mca + + reg: + items: + - description: Register region of the MCA clusters proper + - description: Register region of the DMA glue and its FIFOs + + interrupts: + minItems: 4 + maxItems: 6 + description: + One interrupt per each cluster + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + + dmas: + minItems: 16 + maxItems: 24 + description: + DMA channels corresponding to the SERDES units in the peripheral. They are + listed in groups of four per cluster, and within the group they are given + as associated to the TXA, RXA, TXB, RXB units. + + dma-names: + minItems: 16 + items: + - const: tx0a + - const: rx0a + - const: tx0b + - const: rx0b + - const: tx1a + - const: rx1a + - const: tx1b + - const: rx1b + - const: tx2a + - const: rx2a + - const: tx2b + - const: rx2b + - const: tx3a + - const: rx3a + - const: tx3b + - const: rx3b + - const: tx4a + - const: rx4a + - const: tx4b + - const: rx4b + - const: tx5a + - const: rx5a + - const: tx5b + - const: rx5b + description: | + Names for the DMA channels: 'tx'/'rx', then cluster number, then 'a'/'b' + based on the associated SERDES unit. + + clocks: + minItems: 4 + maxItems: 6 + description: + Clusters' input reference clock. + + resets: + maxItems: 1 + + power-domains: + minItems: 5 + maxItems: 7 + description: + First a general power domain for register access, then the power + domains of individual clusters for their operation. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - power-domains + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + mca: i2s@9b600000 { + compatible = "apple,t6000-mca", "apple,mca"; + reg = <0x9b600000 0x10000>, + <0x9b200000 0x20000>; + + clocks = <&nco 0>, <&nco 1>, <&nco 2>, <&nco 3>; + power-domains = <&ps_audio_p>, <&ps_mca0>, <&ps_mca1>, + <&ps_mca2>, <&ps_mca3>; + dmas = <&admac 0>, <&admac 1>, <&admac 2>, <&admac 3>, + <&admac 4>, <&admac 5>, <&admac 6>, <&admac 7>, + <&admac 8>, <&admac 9>, <&admac 10>, <&admac 11>, + <&admac 12>, <&admac 13>, <&admac 14>, <&admac 15>; + dma-names = "tx0a", "rx0a", "tx0b", "rx0b", + "tx1a", "rx1a", "tx1b", "rx1b", + "tx2a", "rx2a", "tx2b", "rx2b", + "tx3a", "rx3a", "tx3b", "rx3b"; + + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/arm,pl041.yaml b/Documentation/devicetree/bindings/sound/arm,pl041.yaml new file mode 100644 index 000000000..7896b8150 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arm,pl041.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/arm,pl041.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Arm Ltd. PrimeCell PL041 AACI sound interface + +maintainers: + - Andre Przywara <andre.przywara@arm.com> + +description: + The Arm PrimeCell Advanced Audio CODEC Interface (AACI) is an AMBA compliant + peripheral that provides communication with an audio CODEC using the AC-link + protocol. + +# We need a select here so we don't match all nodes with 'arm,primecell' +select: + properties: + compatible: + contains: + const: arm,pl041 + required: + - compatible + +properties: + compatible: + items: + - const: arm,pl041 + - const: arm,primecell + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + description: APB register access clock + + clock-names: + const: apb_pclk + +required: + - compatible + - reg + - interrupts + - clocks + +additionalProperties: false + +examples: + - | + audio-controller@40000 { + compatible = "arm,pl041", "arm,primecell"; + reg = <0x040000 0x1000>; + interrupts = <11>; + clocks = <&v2m_clk24mhz>; + clock-names = "apb_pclk"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt new file mode 100644 index 000000000..953c092db --- /dev/null +++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt @@ -0,0 +1,26 @@ +Device Tree bindings for the Armada 370 DB audio +================================================ + +These Device Tree bindings are used to describe the audio complex +found on the Armada 370 DB platform. + +Mandatory properties: + + * compatible: must be "marvell,a370db-audio" + + * marvell,audio-controller: a phandle that points to the audio + controller of the Armada 370 SoC. + + * marvell,audio-codec: a set of three phandles that points to: + + 1/ the analog audio codec connected to the Armada 370 SoC + 2/ the S/PDIF transceiver + 3/ the S/PDIF receiver + +Example: + + sound { + compatible = "marvell,a370db-audio"; + marvell,audio-controller = <&audio_controller>; + marvell,audio-codec = <&audio_codec &spdif_out &spdif_in>; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4104.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4104.yaml new file mode 100644 index 000000000..86f6061d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4104.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/asahi-kasei,ak4104.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4104 S/PDIF transmitter + +allOf: + - $ref: dai-common.yaml# + +maintainers: + - Daniel Mack <github@zonque.org> + - Xiaxi Shen <shenxiaxi26@gmail.com> + +properties: + compatible: + const: asahi-kasei,ak4104 + + reg: + description: Chip select number on the SPI bus + maxItems: 1 + + vdd-supply: + description: A regulator node providing between 2.7V and 3.6V. + + reset-gpios: + maxItems: 1 + description: Optional GPIO spec for the reset pin, deasserted + before communication starts. + +required: + - compatible + - reg + - vdd-supply + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@0 { + compatible = "asahi-kasei,ak4104"; + reg = <0>; + vdd-supply = <&vdd_3v3_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4375.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4375.yaml new file mode 100644 index 000000000..bc07fcba5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4375.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4375 DAC and headphones amplifier + +maintainers: + - Vincent Knecht <vincent.knecht@mailoo.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: asahi-kasei,ak4375 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + avdd-supply: + description: regulator phandle for the AVDD power supply. + + tvdd-supply: + description: regulator phandle for the TVDD power supply. + + pdn-gpios: + description: optional GPIO to set the PDN pin. + +required: + - compatible + - reg + - '#sound-dai-cells' + - avdd-supply + - tvdd-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + headphones: audio-codec@10 { + compatible = "asahi-kasei,ak4375"; + reg = <0x10>; + avdd-supply = <®_headphones_avdd>; + tvdd-supply = <&pm8916_l6>; + pdn-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphones_pdn_default>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml new file mode 100644 index 000000000..3a3313ea0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + "#sound-dai-cells": + const: 0 + + 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: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: asahi-kasei,ak4458 + + then: + properties: + dsd-path: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@10 { + compatible = "asahi-kasei,ak4458"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4554.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4554.yaml new file mode 100644 index 000000000..c77d85df2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4554.yaml @@ -0,0 +1,27 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4554.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4554 sound codec + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + - Liam Girdwood <lgirdwood@gmail.com> + - Mark Brown <broonie@kernel.org> + +properties: + compatible: + const: asahi-kasei,ak4554 + +required: + - compatible + +additionalProperties: false + +examples: + - | + codec { + compatible = "asahi-kasei,ak4554"; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4613.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4613.yaml new file mode 100644 index 000000000..b49a6cff9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4613.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + ports: + $ref: audio-graph-port.yaml#/definitions/ports + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: codec@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4619.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4619.yaml new file mode 100644 index 000000000..d412531ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4619.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4619.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4619 I2C transmitter + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + - Khanh Le <khanh.le.xr@renesas.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: asahi-kasei,ak4619 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@10 { + compatible = "asahi-kasei,ak4619"; + reg = <0x10>; + + clocks = <&rcar_sound>; + clock-names = "mclk"; + + #sound-dai-cells = <0>; + port { + ak4619_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4642.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4642.yaml new file mode 100644 index 000000000..fc03f0373 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4642.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - asahi-kasei,ak4642 + - asahi-kasei,ak4643 + - asahi-kasei,ak4648 + + reg: + maxItems: 1 + + "#clock-cells": + const: 0 + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-frequency: + description: common clock binding; frequency of MCKO + + clock-output-names: + description: common clock name + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + #sound-dai-cells = <0>; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak5558.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak5558.yaml new file mode 100644 index 000000000..18919d911 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak5558.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak5558: codec@10 { + compatible = "asahi-kasei,ak5558"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml b/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml new file mode 100644 index 000000000..89a67f8e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,asoc-wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel wm8904 audio codec complex + +maintainers: + - Dharma Balasubiramani <dharma.b@microchip.com> + +description: + The ASoC audio complex configuration for Atmel with WM8904 audio codec. + +properties: + compatible: + const: atmel,asoc-wm8904 + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8731 audio codec. + + atmel,audio-routing: + description: + A list of the connections between audio components. Each entry is a pair + of strings, the first being the connection's sink, the second being the + connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + - Mic + # WM8904 CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - MICBIAS + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,at91-ssc.yaml b/Documentation/devicetree/bindings/sound/atmel,at91-ssc.yaml new file mode 100644 index 000000000..ce99c2d8c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,at91-ssc.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,at91-ssc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel Serial Synchronous Serial (SSC) + +maintainers: + - Andrei Simion <andrei.simion@microchip.com> + +description: + The Atmel Synchronous Serial Controller (SSC) provides a versatile + synchronous communication link for audio and telecom applications, + supporting protocols like I2S, Short Frame Sync, and Long Frame Sync. + +properties: + compatible: + oneOf: + - enum: + - atmel,at91rm9200-ssc + - atmel,at91sam9g45-ssc + - items: + - enum: + - microchip,sam9x7-ssc + - const: atmel,at91sam9g45-ssc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: pclk + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + atmel,clk-from-rk-pin: + description: + Specify the clock source for the SSC (Synchronous Serial Controller) + when operating in slave mode. By default, the clock is sourced from + the TK pin. + type: boolean + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - atmel,at91sam9g45-ssc + then: + required: + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/irq.h> + + ssc@100000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0x100000 0x4000>; + interrupts = <28 IRQ_TYPE_LEVEL_HIGH 5>; + dmas = <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(38))>, + <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(39))>; + dma-names = "tx", "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 28>; + clock-names = "pclk"; + #sound-dai-cells = <0>; + }; + + ssc@c00000 { + compatible = "atmel,at91rm9200-ssc"; + reg = <0xc00000 0x4000>; + interrupts = <14 IRQ_TYPE_LEVEL_HIGH 5>; + clocks = <&pmc PMC_TYPE_PERIPHERAL 14>; + clock-names = "pclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml new file mode 100644 index 000000000..627da2d89 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,at91sam9g20ek-wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9g20ek wm8731 audio complex + +maintainers: + - Balakrishnan Sambath <balakrishnan.s@microchip.com> + +description: + The audio complex configuration for Atmel at91sam9g20ek with WM8731 audio codec. + +properties: + compatible: + const: atmel,at91sam9g20ek-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: A list of the connections between audio components. + minItems: 2 + maxItems: 4 + items: + enum: + # Board Connectors + - Ext Spk + - Int Mic + + # CODEC Pins + - LOUT + - ROUT + - LHPOUT + - RHPOUT + - LLINEIN + - RLINEIN + - MICIN + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of WM8731 audio codec. + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,at91sam9g20ek-wm8731-audio"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + atmel,model = "wm8731 @ AT91SAMG20EK"; + atmel,audio-routing = + "Ext Spk", "LHPOUT", + "Int Mic", "MICIN"; + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml b/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml new file mode 100644 index 000000000..33717b728 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sam9x5-wm8731-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9x5ek wm8731 audio complex + +maintainers: + - Dharma Balasubiramani <dharma.b@microchip.com> + +description: + The audio complex configuration for Atmel at91sam9x5ek with WM8731 audio codec. + +properties: + compatible: + const: atmel,sam9x5-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8731 audio codec. + + atmel,audio-routing: + description: + A list of the connections between audio components. Each entry is a pair + of strings, the first being the connection's sink, the second being the + connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + + # CODEC Pins + - LOUT + - ROUT + - LHPOUT + - RHPOUT + - LLINEIN + - RLINEIN + - MICIN + +required: + - compatible + - atmel,model + - atmel,ssc-controller + - atmel,audio-codec + - atmel,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,sam9x5-wm8731-audio"; + + atmel,model = "wm8731 @ AT91SAM9X5EK"; + + atmel,audio-routing = + "Headphone Jack", "RHPOUT", + "Headphone Jack", "LHPOUT", + "LLINEIN", "Line In Jack", + "RLINEIN", "Line In Jack"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml new file mode 100644 index 000000000..ae3162fcf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-classd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel ClassD Amplifier + +maintainers: + - Nicolas Ferre <nicolas.ferre@microchip.com> + - Alexandre Belloni <alexandre.belloni@bootlin.com> + - Claudiu Beznea <claudiu.beznea@microchip.com> + +description: + The Audio Class D Amplifier (CLASSD) is a digital input, Pulse Width + Modulated (PWM) output stereo Class D amplifier. + +properties: + compatible: + oneOf: + - items: + - const: atmel,sama5d2-classd + - items: + - const: microchip,sam9x7-classd + - const: atmel,sama5d2-classd + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + const: tx + + clocks: + maxItems: 2 + + clock-names: + items: + - const: pclk + - const: gclk + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + default: CLASSD + description: The user-visible name of this sound complex. + + atmel,pwm-type: + $ref: /schemas/types.yaml#/definitions/string + enum: + - single + - diff + default: single + description: PWM modulation type. + + atmel,non-overlap-time: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 5 + - 10 + - 15 + - 20 + default: 10 + description: + Set non-overlapping time, the unit is nanosecond(ns). + Non-overlapping will be disabled if not specified. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clock-names + - clocks + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + classd: sound@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>; + clock-names = "pclk", "gclk"; + assigned-clocks = <&classd_gclk>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml new file mode 100644 index 000000000..0cd1ff89b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel I2S controller + +maintainers: + - Nicolas Ferre <nicolas.ferre@microchip.com> + - Alexandre Belloni <alexandre.belloni@bootlin.com> + - Claudiu Beznea <claudiu.beznea@microchip.com> + +description: + Atmel I2S (Inter-IC Sound Controller) bus is the standard + interface for connecting audio devices, such as audio codecs. + +properties: + compatible: + const: atmel,sama5d2-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + - description: Generated clock (Optional) + - description: I2S mux clock (Optional). Set + with gclk when Master Mode is required. + minItems: 1 + + clock-names: + items: + - const: pclk + - const: gclk + - const: muxclk + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f8050000 { + compatible = "atmel,sama5d2-i2s"; + reg = <0xf8050000 0x300>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(31))>, + <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(32))>; + dma-names = "tx", "rx"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s0_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml new file mode 100644 index 000000000..f320b561f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-pdmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel PDMIC decoder + +maintainers: + - Claudiu Beznea <claudiu.beznea@microchip.com> + +description: + Atmel Pulse Density Modulation Interface Controller + (PDMIC) peripheral is a mono PDM decoder module + that decodes an incoming PDM sample stream. + +properties: + compatible: + const: atmel,sama5d2-pdmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: peripheral clock + - description: generated clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + maxItems: 1 + + dma-names: + const: rx + + atmel,mic-min-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The minimal frequency that the microphone supports. + + atmel,mic-max-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The maximal frequency that the microphone supports. + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + default: PDMIC + description: The user-visible name of this sound card. + + atmel,mic-offset: + $ref: /schemas/types.yaml#/definitions/int32 + default: 0 + description: The offset that should be added. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clock-names + - clocks + - atmel,mic-min-freq + - atmel,mic-max-freq + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + pdmic: sound@f8018000 { + compatible = "atmel,sama5d2-pdmic"; + reg = <0xf8018000 0x124>; + interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(50))>; + dma-names = "rx"; + clocks = <&pdmic_clk>, <&pdmic_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pdmic_default>; + atmel,model = "PDMIC@sama5d2_xplained"; + atmel,mic-min-freq = <1000000>; + atmel,mic-max-freq = <3246000>; + atmel,mic-offset = <0x0>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel_ac97c.txt b/Documentation/devicetree/bindings/sound/atmel_ac97c.txt new file mode 100644 index 000000000..b151bd902 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel_ac97c.txt @@ -0,0 +1,20 @@ +* Atmel AC97 controller + +Required properties: + - compatible: "atmel,at91sam9263-ac97c" + - reg: Address and length of the register set for the device + - interrupts: Should contain AC97 interrupt + - ac97-gpios: Please refer to soc-ac97link.txt, only ac97-reset is used +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound@fffa0000 { + compatible = "atmel,at91sam9263-ac97c"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ac97>; + reg = <0xfffa0000 0x4000>; + interrupts = <18 IRQ_TYPE_LEVEL_HIGH 5>; + + ac97-gpios = <&pioB 0 0 &pioB 2 0 &pioC 29 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml new file mode 100644 index 000000000..274092ef3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +allOf: + - $ref: /schemas/sound/audio-graph.yaml# + +properties: + compatible: + enum: + - audio-graph-card + - audio-graph-scu-card + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "audio-graph-card"; + + dais = <&cpu_port_a>; + }; + + cpu { + /* + * dai-controller own settings + */ + + port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + dai-format = "left_j"; + }; + }; + }; + + codec { + /* + * codec own settings + */ + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml new file mode 100644 index 000000000..40eb1d7d6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + $ref: audio-graph.yaml#/properties/routing + 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 + multi: + type: object + description: Multi-CPU/Codec node + dpcm: + type: object + description: DPCM node + codec2codec: + type: object + description: Codec to Codec node + hp-det-gpios: + $ref: audio-graph.yaml#/properties/hp-det-gpios + mic-det-gpios: + $ref: audio-graph.yaml#/properties/mic-det-gpios + widgets: + $ref: audio-graph.yaml#/properties/widgets + +required: + - compatible + - links + +additionalProperties: false + +... diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml new file mode 100644 index 000000000..d1cbfc5ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -0,0 +1,131 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-port.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card 'port' + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +definitions: + port-base: + allOf: + - $ref: /schemas/graph.yaml#/$defs/port-base + - $ref: /schemas/sound/dai-params.yaml# + properties: + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + playback-only: + description: port connection used only for playback + $ref: /schemas/types.yaml#/definitions/flag + capture-only: + description: port connection used only for capture + $ref: /schemas/types.yaml#/definitions/flag + link-trigger-order: + description: trigger order for both start/stop + $ref: /schemas/types.yaml#/definitions/uint32-array + link-trigger-order-start: + description: trigger order for start + $ref: /schemas/types.yaml#/definitions/uint32-array + link-trigger-order-stop: + description: trigger order for stop + $ref: /schemas/types.yaml#/definitions/uint32-array + + endpoint-base: + allOf: + - $ref: /schemas/graph.yaml#/$defs/endpoint-base + - $ref: /schemas/sound/dai-params.yaml# + properties: + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + frame-master: + description: Indicates dai-link frame master. + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + bitclock-master: + description: Indicates dai-link bit clock master + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + clocks: + description: Indicates system clock + maxItems: 1 + system-clock-frequency: + $ref: simple-card.yaml#/definitions/system-clock-frequency + system-clock-direction-out: + $ref: simple-card.yaml#/definitions/system-clock-direction-out + system-clock-fixed: + $ref: simple-card.yaml#/definitions/system-clock-fixed + + dai-format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai-tdm-slot-num: + description: Number of slots in use. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: Width in bits for each slot. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width-map: + description: Mapping of sample widths to slot widths. For hardware + that cannot support a fixed slot width or a slot width always + equal to sample width. A matrix of one or more 3-tuples. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - + description: Sample width in bits + minimum: 8 + maximum: 64 + - + description: Slot width in bits + minimum: 8 + maximum: 256 + - + description: Slot count + minimum: 1 + maximum: 64 + + ports: + $ref: "#/definitions/port-base" + unevaluatedProperties: false + patternProperties: + "^port(@[0-9a-f]+)?$": + $ref: "#/definitions/port-base" + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: "#/definitions/endpoint-base" + unevaluatedProperties: false + +allOf: + - $ref: "#/definitions/port-base" + +patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: "#/definitions/endpoint-base" + unevaluatedProperties: false + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml new file mode 100644 index 000000000..9899d9d19 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -0,0 +1,53 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pa-gpios: + maxItems: 1 + hp-det-gpio: + deprecated: true + maxItems: 1 + hp-det-gpios: + maxItems: 1 + mic-det-gpio: + deprecated: true + maxItems: 1 + mic-det-gpios: + maxItems: 1 + +required: + - dais + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml b/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml new file mode 100644 index 000000000..d3cc1ea4a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-iio-aux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio IIO auxiliary + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: + Auxiliary device based on Industrial I/O device channels + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: audio-iio-aux + + io-channels: + description: + Industrial I/O device channels used + + io-channel-names: + description: + Industrial I/O channel names related to io-channels. + These names are used to provides sound controls, widgets and routes names. + + snd-control-invert-range: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + A list of 0/1 flags defining whether or not the related channel is + inverted + items: + enum: [0, 1] + default: 0 + description: | + Invert the sound control value compared to the IIO channel raw value. + - 1: The related sound control value is inverted meaning that the + minimum sound control value correspond to the maximum IIO channel + raw value and the maximum sound control value correspond to the + minimum IIO channel raw value. + - 0: The related sound control value is not inverted meaning that the + minimum (resp maximum) sound control value correspond to the + minimum (resp maximum) IIO channel raw value. + +required: + - compatible + - io-channels + - io-channel-names + +unevaluatedProperties: false + +examples: + - | + iio-aux { + compatible = "audio-iio-aux"; + io-channels = <&iio 0>, <&iio 1>, <&iio 2>, <&iio 3>; + io-channel-names = "CH0", "CH1", "CH2", "CH3"; + /* Invert CH1 and CH2 */ + snd-control-invert-range = <0 1 1 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml new file mode 100644 index 000000000..bc6c6b172 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw8738.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw8738.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic AW8738 Audio Amplifier + +maintainers: + - Stephan Gerhold <stephan@gerhold.net> + +description: + The Awinic AW8738 is a simple audio amplifier with different operation modes + (set using one-wire pulse control). The mode configures the speaker-guard + function (primarily the power limit for the amplifier). + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: awinic,aw8738 + + mode-gpios: + description: + GPIO used for one-wire pulse control. The pin is typically called SHDN + (active-low), but this is misleading since it is actually more than + just a simple shutdown/enable control. + maxItems: 1 + + awinic,mode: + description: Operation mode (number of pulses for one-wire pulse control) + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 7 + + sound-name-prefix: true + +required: + - compatible + - mode-gpios + - awinic,mode + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + audio-amplifier { + compatible = "awinic,aw8738"; + mode-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + awinic,mode = <5>; + sound-name-prefix = "Speaker Amp"; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw87390.yaml b/Documentation/devicetree/bindings/sound/awinic,aw87390.yaml new file mode 100644 index 000000000..9c1baae76 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw87390.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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. + +properties: + compatible: + oneOf: + - enum: + - awinic,aw87390 + - items: + - enum: + - anbernic,rgds-amp + - const: awinic,aw87391 + + 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" + +unevaluatedProperties: false + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - awinic,aw87390 + then: + required: + - awinic,audio-channel + + - if: + properties: + compatible: + contains: + enum: + - anbernic,rgds-amp + then: + properties: + vdd-supply: true + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@58 { + compatible = "awinic,aw87390"; + reg = <0x58>; + #sound-dai-cells = <0>; + awinic,audio-channel = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml new file mode 100644 index 000000000..b9abb1094 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml @@ -0,0 +1,102 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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,aw88081 + - awinic,aw88083 + - awinic,aw88166 + - awinic,aw88261 + - awinic,aw88395 + - awinic,aw88399 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + reset-gpios: + maxItems: 1 + + dvdd-supply: true + + firmware-name: + maxItems: 1 + description: Name of the *_acf.bin file used for amplifier initialization + + 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,aw88081 + - awinic,aw88083 + - awinic,aw88261 + then: + properties: + reset-gpios: false + - if: + properties: + compatible: + contains: + const: awinic,aw88261 + then: + required: + - dvdd-supply + else: + properties: + dvdd-supply: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@34 { + compatible = "awinic,aw88395"; + reg = <0x34>; + #sound-dai-cells = <0>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + awinic,audio-channel = <0>; + awinic,sync-flag; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt new file mode 100644 index 000000000..b6cc5f6f7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt @@ -0,0 +1,92 @@ +Devicetree bindings for the Axentia TSE-850 audio complex + +Required properties: + - compatible: "axentia,tse850-pcm5142" + - axentia,cpu-dai: The phandle of the cpu dai. + - axentia,audio-codec: The phandle of the PCM5142 codec. + - axentia,add-gpios: gpio specifier that controls the mixer. + - axentia,loop1-gpios: gpio specifier that controls loop relays on channel 1. + - axentia,loop2-gpios: gpio specifier that controls loop relays on channel 2. + - axentia,ana-supply: Regulator that supplies the output amplifier. Must + support voltages in the 2V - 20V range, in 1V steps. + +The schematics explaining the gpios are as follows: + + loop1 relays + IN1 +---o +------------+ o---+ OUT1 + \ / + + + + | / | + +--o +--. | + | add | | + | V | + | .---. | + DAC +----------->|Sum|---+ + | '---' | + | | + + + + + IN2 +---o--+------------+--o---+ OUT2 + loop2 relays + +The 'loop1' gpio pin 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/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.yaml b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.yaml new file mode 100644 index 000000000..f3cfe9268 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.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/brcm,bcm2835-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Broadcom BCM2835 SoC I2S/PCM module + +maintainers: + - Florian Fainelli <florian.fainelli@broadcom.com> + +properties: + compatible: + const: brcm,bcm2835-i2s + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + dmas: + items: + - description: Transmission DMA controller phandle and request line. + - description: Reception DMA controller phandle and request line. + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - reg + - clocks + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/bcm2835.h> + + i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = <0x7e203000 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; + dmas = <&dma 2>, <&dma 3>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt new file mode 100644 index 000000000..007f524b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt @@ -0,0 +1,29 @@ +Broadcom DSL/PON BCM63xx Audio I2S controller + +Required properties: +- compatible: Should be "brcm,bcm63xx-i2s". +- #address-cells: 32bit valued, 1 cell. +- #size-cells: 32bit valued, 0 cell. +- reg: Should contain audio registers location and length +- interrupts: Should contain the interrupt for the controller. +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: One of each entry matching the clocks phandles list: + - "i2sclk" (generated clock) Required. + - "i2sosc" (fixed 200MHz clock) Required. + +(1) : The generated clock is required only when any of TX and RX + works on Master Mode. +(2) : The fixed 200MHz clock is from internal chip and always on + +Example: + + i2s: bcm63xx-i2s { + #address-cells = <1>; + #size-cells = <0>; + compatible = "brcm,bcm63xx-i2s"; + reg = <0xFF802080 0xFF>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&i2sclk>, <&osc>; + clock-names = "i2sclk","i2sosc"; + }; diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt new file mode 100644 index 000000000..630bf7c03 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt @@ -0,0 +1,63 @@ +BROADCOM Cygnus Audio I2S/TDM/SPDIF controller + +Required properties: + - compatible : "brcm,cygnus-audio" + - #address-cells: 32bit valued, 1 cell. + - #size-cells: 32bit valued, 0 cell. + - reg : Should contain audio registers location and length + - reg-names: names of the registers listed in "reg" property + Valid names are "aud" and "i2s_in". "aud" contains a + set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains + a set of I2S_IN registers. + - clocks: PLL and leaf clocks used by audio ports + - assigned-clocks: PLL and leaf clocks + - assigned-clock-parents: parent clocks of the assigned clocks + (usually the PLL) + - assigned-clock-rates: List of clock frequencies of the + assigned clocks + - clock-names: names of 3 leaf clocks used by audio ports + Valid names are "ch0_audio", "ch1_audio", "ch2_audio" + - interrupts: audio DMA interrupt number + +SSP Subnode properties: +- reg: The index of ssp port interface to use + Valid value are 0, 1, 2, or 3 (for spdif) + +Example: + cygnus_audio: audio@180ae000 { + compatible = "brcm,cygnus-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>; + reg-names = "aud", "i2s_in"; + clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>; + assigned-clock-rates = <1769470191>, + <0>, + <0>, + <0>; + clock-names = "ch0_audio", "ch1_audio", "ch2_audio"; + interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>; + + ssp0: ssp_port@0 { + reg = <0>; + }; + + ssp1: ssp_port@1 { + reg = <1>; + }; + + ssp2: ssp_port@2 { + reg = <2>; + }; + + spdif: spdif_port@3 { + reg = <3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.yaml b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.yaml new file mode 100644 index 000000000..9617acef3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.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/cdns,xtfpga-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: I2S controller built into xtfpga Xtensa bitstreams + +maintainers: + - Max Filippov <jcmvbkbc@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: cdns,xtfpga-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + description: phandle to the clk used as master clock. I2S bus clock is derived from it. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + +unevaluatedProperties: false + +examples: + - | + i2s@d080000 { + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l36.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l36.yaml new file mode 100644 index 000000000..2a142b32a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l36.yaml @@ -0,0 +1,240 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs35l36.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS35L36 Speaker Amplifier + +maintainers: + - David Rhodes <drhodes@opensource.cirrus.com> + - patches@opensource.cirrus.com + +description: + CS35L36 is a boosted mono Class D amplifier + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,cs35l36 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + VA-supply: + description: Voltage regulator of analog internal section + + VP-supply: + description: Voltage regulator of boost converter + + reset-gpios: + maxItems: 1 + + cirrus,boost-ctl-millivolt: + description: Boost converter output voltage (step 50) + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 2550 + maximum: 12000 + + cirrus,boost-peak-milliamp: + description: Boost-converter peak current limit (step 50) + $ref: /schemas/types.yaml#/definitions/uint32 + default: 4500 + minimum: 1600 + maximum: 4500 + + cirrus,boost-ind-nanohenry: + description: Initial inductor estimation reference value (1000=1μH, 1200=1.2μH) + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1000 + + cirrus,multi-amp-mode: + description: Hi-Z ASP port when more than one amplifier in system + type: boolean + + cirrus,boost-ctl-select: + description: Boost converter control source selection + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: + - 0 # Control Port + - 1 # Class + - 2 # Sync + + cirrus,amp-pcm-inv: + description: Invert incoming PCM data when true + type: boolean + + cirrus,imon-pol-inv: + description: Invert polarity of outbound IMON feedback when true + type: boolean + + cirrus,vmon-pol-inv: + description: Invert polarity of outbound VMON feedback when true + type: boolean + + cirrus,dcm-mode-enable: + description: Enable boost converter automatic Discontinuous Conduction Mode + type: boolean + + cirrus,weak-fet-disable: + description: Reduce output driver strength in Weak-FET Drive Mode when true + type: boolean + + cirrus,classh-wk-fet-delay-ms: + description: Weak-FET entry delay + default: 100 + enum: [0, 5, 10, 50, 100, 200, 500, 1000] + + cirrus,classh-weak-fet-thld-millivolt: + description: Weak-FET drive threshold + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [50, 100, 150, 200, 250, 300, 350, 400, 450, 500, 550, 600, 650, 700] + + cirrus,temp-warn-threshold: + description: Overtemperature warning threshold + $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + enum: + - 0 # 105°C + - 1 # 115°C + - 2 # 125°C + - 3 # 135°C + + cirrus,irq-drive-select: + description: Interrupt output driver type + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: + - 0 # open-drain + - 1 # push-pull + + cirrus,irq-gpio-select: + description: Programmable IRQ pin selection + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # PDM_DATA/SWIRE_SD/INT + - 1 # GPIO + + cirrus,vpbr-config: + $ref: "#/$defs/vpbr-config" + +$defs: + vpbr-config: + description: Brownout prevention configuration sub-node + type: object + additionalProperties: false + + properties: + cirrus,vpbr-en: + description: VBST brownout prevention enable + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + enum: + - 0 # disabled + - 1 # enabled + + cirrus,vpbr-thld: + description: Initial VPBR threshold voltage + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 2 + maximum: 31 + + cirrus,vpbr-atk-rate: + description: Attenuation attack step rate + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + + cirrus,vpbr-atk-vol: + description: VP brownout prevention step size + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + + cirrus,vpbr-max-attn: + description: Maximum attenuation during VP brownout prevention (dB) + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + + cirrus,vpbr-wait: + description: Delay between brownout clearance and attenuation release (ms) + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: + - 0 # 10 + - 1 # 100 + - 2 # 250 + - 3 # 500 + + cirrus,vpbr-rel-rate: + description: Attenuation release step rate + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + + cirrus,vpbr-mute-en: + description: Mute audio if maximum attenuation reached + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + +required: + - compatible + - reg + - interrupts + - VA-supply + - cirrus,boost-ctl-millivolt + - cirrus,boost-peak-milliamp + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 GPIO_ACTIVE_HIGH>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0>; + cirrus,classh-wk-fet-delay-ms = <100>; + cirrus,classh-weak-fet-thld-millivolt = <100>; + cirrus,temp-warn-threshold = <1>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <1>; + cirrus,irq-gpio-select = <1>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0>; + cirrus,vpbr-thld = <5>; + cirrus,vpbr-atk-rate = <2>; + cirrus,vpbr-atk-vol = <1>; + cirrus,vpbr-max-attn = <9>; + cirrus,vpbr-wait = <1>; + cirrus,vpbr-rel-rate = <5>; + cirrus,vpbr-mute-en = <0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml new file mode 100644 index 000000000..e6cf2ebcd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -0,0 +1,215 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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 + + cirrus,subsystem-id: + $ref: /schemas/types.yaml#/definitions/string + description: + Subsystem ID. If this property is present, it sets the system name, + used to identify the firmware and tuning to load. + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + cirrus,boost-type: + const: 0 + then: + required: + - cirrus,boost-peak-milliamp + - cirrus,boost-ind-nanohenry + - cirrus,boost-cap-microfarad + else: + if: + properties: + cirrus,boost-type: + const: 1 + then: + required: + - cirrus,gpio1-output-enable + - cirrus,gpio1-src-select + properties: + cirrus,boost-peak-milliamp: false + cirrus,boost-ind-nanohenry: false + cirrus,boost-cap-microfarad: false + cirrus,gpio1-src-select: + enum: [1] + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l41: speaker-amp@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 GPIO_ACTIVE_HIGH>; + + cirrus,boost-type = <0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml new file mode 100644 index 000000000..70f6c62ae --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml @@ -0,0 +1,159 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs35l45.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS35L45 Speaker Amplifier + +maintainers: + - Ricardo Rivera-Matos <rriveram@opensource.cirrus.com> + - Richard Fitzgerald <rf@opensource.cirrus.com> + +description: | + CS35L45 is a Boosted Mono Class D Amplifier with DSP + Speaker Protection and Adaptive Battery Management. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,cs35l45 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + reset-gpios: + maxItems: 1 + + vdd-a-supply: + description: voltage regulator phandle for the VDD_A supply + + vdd-batt-supply: + description: voltage regulator phandle for the VDD_BATT supply + + spi-max-frequency: + maximum: 5000000 + + cirrus,asp-sdout-hiz-ctrl: + description: + Audio serial port SDOUT Hi-Z control. Sets the Hi-Z + configuration for SDOUT pin of amplifier. Logical OR of + CS35L45_ASP_TX_HIZ_xxx values. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 + default: 2 + +patternProperties: + "^cirrus,gpio-ctrl[1-3]$": + description: + GPIO pins configuration. + type: object + additionalProperties: false + properties: + gpio-dir: + description: + GPIO pin direction. Valid only when 'gpio-ctrl' is 1 + 0 = Output + 1 = Input + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 1 + gpio-lvl: + description: + GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 + 0 = Low + 1 = High + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 0 + gpio-op-cfg: + description: + GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 + 0 = CMOS + 1 = Open Drain + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 0 + gpio-pol: + description: + GPIO output polarity select. Valid only when 'gpio-ctrl' is 1 + and 'gpio-dir' is 0 + 0 = Non-inverted, Active High + 1 = Inverted, Active Low + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + default: 0 + gpio-ctrl: + description: + Defines the function of the GPIO pin. + GPIO1 + 0 = High impedance input + 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + 2 = Pin acts as MDSYNC, direction controlled by MDSYNC + 3-7 = Reserved + GPIO2 + 0 = High impedance input + 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + 2 = Pin acts as open drain INT + 3 = Reserved + 4 = Pin acts as push-pull output INT. Active low. + 5 = Pin acts as push-pull output INT. Active high. + 6,7 = Reserved + GPIO3 + 0 = High impedance input + 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + 2-7 = Reserved + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + default: 0 +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/cs35l45.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l45: cs35l45@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l45"; + reg = <2>; + spi-max-frequency = <5000000>; + vdd-a-supply = <&dummy_vreg>; + vdd-batt-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + cirrus,asp-sdout-hiz-ctrl = <(CS35L45_ASP_TX_HIZ_UNUSED | + CS35L45_ASP_TX_HIZ_DISABLED)>; + cirrus,gpio-ctrl1 { + gpio-ctrl = <0x2>; + }; + cirrus,gpio-ctrl2 { + gpio-ctrl = <0x2>; + }; + cirrus,gpio-ctrl3 { + gpio-ctrl = <0x1>; + gpio-dir = <0x1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml new file mode 100644 index 000000000..156560b2a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs4234.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic cs4234 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: + The CS4234 is a highly versatile CODEC that combines 4 channels of + high performance analog to digital conversion, 4 channels of high + performance digital to analog conversion for audio, and 1 channel of + digital to analog conversion to provide a nondelayed audio reference + signal to an external Class H tracking power supply. If not used to + drive a tracking power supply, the 5th DAC can instead be used as a + standard audio grade DAC, with performance specifications identical + to that of the 4 DACs in the audio path. Additionally, the CS4234 + includes tunable group delay for each of the 4 audio DAC paths to + provide lead time for the external switch-mode power supply, and a + nondelayed path into the DAC outputs for input signals requiring a + low-latency path to the outputs. + +properties: + compatible: + enum: + - cirrus,cs4234 + + reg: + description: + The 7-bit I2C address depends on the state of the ADx pins, in + binary given by [0 0 1 0 AD2 AD1 AD0 0]. + items: + minimum: 0x10 + maximum: 0x17 + + VA-supply: + description: + Analogue power supply. + + VL-supply: + description: + Interface power supply. + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - VA-supply + - VL-supply + +additionalProperties: false + +examples: + - | + i2c@e0004000 { + #address-cells = <1>; + #size-cells = <0>; + reg = <0xe0004000 0x1000>; + + cs4234: codec@11 { + compatible = "cirrus,cs4234"; + reg = <0x11>; + + VA-supply = <&vdd3v3>; + VL-supply = <&vdd3v3>; + + reset-gpios = <&gpio 0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml new file mode 100644 index 000000000..336e11773 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4270.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/cirrus,cs4270.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS4270 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: + The CS4270 is a stereo audio codec. The driver for this device currently only + supports I2C. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: cirrus,cs4270 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + reset-gpios: + description: + This pin will be deasserted before communication to the codec starts. + maxItems: 1 + + va-supply: + description: Analog power supply. + + vd-supply: + description: Digital power supply. + + vlc-supply: + description: Serial Control Port power supply. + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml new file mode 100644 index 000000000..d286eb169 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml @@ -0,0 +1,111 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs4271.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS4271 audio CODEC + +maintainers: + - Alexander Sverdlin <alexander.sverdlin@gmail.com> + - Nikita Shubin <nikita.shubin@maquefel.me> + +description: + The CS4271 is a stereo audio codec. This device supports both the I2C + and the SPI bus. + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +properties: + compatible: + const: cirrus,cs4271 + + reg: + maxItems: 1 + + clocks: + items: + - description: + Master clock connected to the MCLK pin if MCLK is an input (i.e. no + crystal used). + + clock-names: + items: + - const: mclk + + spi-cpha: true + + spi-cpol: true + + '#sound-dai-cells': + const: 0 + + reset-gpios: + description: + This pin will be deasserted before communication to the codec starts. + maxItems: 1 + + va-supply: + description: Analog power supply. + + vd-supply: + description: Digital power supply. + + vl-supply: + description: Serial Control Port power supply. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + cirrus,amuteb-eq-bmutec: + description: + When given, the Codec's AMUTEB=BMUTEC flag is enabled. + type: boolean + + cirrus,enable-soft-reset: + description: | + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + codec@0 { + compatible = "cirrus,cs4271"; + reg = <0>; + #sound-dai-cells = <0>; + spi-max-frequency = <6000000>; + spi-cpol; + spi-cpha; + reset-gpios = <&gpio0 1 GPIO_ACTIVE_LOW>; + port { + endpoint { + remote-endpoint = <&i2s_ep>; + }; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml new file mode 100644 index 000000000..af599d873 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml @@ -0,0 +1,226 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l42.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS42L42 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: + The CS42L42 is a low-power audio codec designed for portable applications. + It provides a high-dynamic range, stereo DAC for audio playback and a mono + high-dynamic-range ADC for audio capture. There is an integrated headset + detection block. + +properties: + compatible: + enum: + - cirrus,cs42l42 + - cirrus,cs42l83 + + reg: + description: + The I2C address of the CS42L42. + maxItems: 1 + + VP-supply: + description: + VP power supply. + + VCP-supply: + description: + Charge pump power supply. + + VD_FILT-supply: + description: + FILT+ power supply. + + VL-supply: + description: + Logic power supply. + + VA-supply: + description: + Analog power supply. + + reset-gpios: + description: + This pin will be asserted and then deasserted to reset the + CS42L42 before communication starts. + maxItems: 1 + + interrupts: + description: + Interrupt for CS42L42 IRQ line. + maxItems: 1 + + cirrus,ts-inv: + description: | + Sets the behaviour of the jack plug detect switch. + + 0 - (Default) Shorted to tip when unplugged, open when plugged. + This is "inverted tip sense (ITS)" in the datasheet. + + 1 - Open when unplugged, shorted to tip when plugged. + This is "normal tip sense (TS)" in the datasheet. + + The CS42L42_TS_INV_* defines are available for this. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 1 + + cirrus,ts-dbnc-rise: + description: | + Debounce the rising edge of TIP_SENSE_PLUG. With no + debounce, the tip sense pin might be noisy on a plug event. + + 0 - 0ms + 1 - 125ms + 2 - 250ms + 3 - 500ms + 4 - 750ms + 5 - 1s (Default) + 6 - 1.25s + 7 - 1.5s + + The CS42L42_TS_DBNCE_* defines are available for this. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + + cirrus,ts-dbnc-fall: + description: | + Debounce the falling edge of TIP_SENSE_UNPLUG. With no + debounce, the tip sense pin might be noisy on an unplug event. + + 0 - 0ms + 1 - 125ms + 2 - 250ms + 3 - 500ms + 4 - 750ms + 5 - 1s (Default) + 6 - 1.25s + 7 - 1.5s + + The CS42L42_TS_DBNCE_* defines are available for this. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 7 + + cirrus,btn-det-init-dbnce: + description: | + This sets how long to wait after enabling button detection + interrupts before servicing button interrupts, to allow the + HS bias time to settle. Value is in milliseconds. + There may be erroneous button interrupts if this debounce time + is too short. + + 0ms - 200ms, + Default = 100ms + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 200 + + cirrus,btn-det-event-dbnce: + description: | + This sets how long to wait after receiving a button press + interrupt before processing it. Allows time for the button + press to make a clean connection with the bias resistors. + Value is in milliseconds. + + 0ms - 20ms, + Default = 10ms + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 20 + + cirrus,bias-lvls: + description: | + For a level-detect headset button scheme, each button will bias + the mic pin to a certain voltage. To determine which button was + pressed, the voltage is compared to sequential, decreasing + voltages, until the compared voltage < bias voltage. + For different hardware setups, a designer might want to tweak this. + This is an array of descending values for the comparator voltage, + given as percent of the HSBIAS voltage. + + Array of 4 values, each 0-63 + < x1 x2 x3 x4 > + Default = < 15 8 4 1 > + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + items: + minimum: 0 + maximum: 63 + + cirrus,hs-bias-ramp-rate: + description: | + If present this sets the rate that the HS bias should rise and fall. + The actual rise and fall times depend on external hardware (the + datasheet gives several rise and fall time examples). + + 0 - Fast rise time; slow, load-dependent fall time + 1 - Fast + 2 - Slow (default) + 3 - Slowest + + The CS42L42_HSBIAS_RAMP_* defines are available for this. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 + + cirrus,hs-bias-sense-disable: + description: | + If present the HSBIAS sense is disabled. Configures HSBIAS output + current sense through the external 2.21-k resistor. HSBIAS_SENSE + is a hardware feature to reduce the potential pop noise when the + headset plug is removed slowly. But on some platforms ESD voltage + will affect it causing plug detection to fail, especially with CTIA + headset type. For different hardware setups, a designer might want + to tweak default behavior. + type: boolean + +required: + - compatible + - reg + - VP-supply + - VCP-supply + - VD_FILT-supply + - VL-supply + - VA-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs42l42.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l42: cs42l42@48 { + compatible = "cirrus,cs42l42"; + reg = <0x48>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD_FILT-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + + reset-gpios = <&axi_gpio_0 1 0>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + + cirrus,ts-inv = <CS42L42_TS_INV_DIS>; + cirrus,ts-dbnc-rise = <CS42L42_TS_DBNCE_1000>; + cirrus,ts-dbnc-fall = <CS42L42_TS_DBNCE_0>; + cirrus,btn-det-init-dbnce = <100>; + cirrus,btn-det-event-dbnce = <10>; + cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>; + cirrus,hs-bias-ramp-rate = <CS42L42_HSBIAS_RAMP_SLOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml new file mode 100644 index 000000000..c1ef9f308 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml @@ -0,0 +1,318 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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. + CS42L43B variant adds dedicated PDM interface, SoundWire Clock Gearing + support and more decimators to ISRCs. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,cs42l43 + - cirrus,cs42l43b + - cirrus,cs42l44 + + 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 + $ref: /schemas/pinctrl/pinctrl.yaml# + additionalProperties: false + + properties: + gpio-controller: true + + "#gpio-cells": + const: 2 + + gpio-ranges: + items: + - items: + - description: A phandle to the CODEC pinctrl node + minimum: 0 + - const: 0 + - const: 0 + - const: 3 + + patternProperties: + "-state$": + oneOf: + - $ref: "#/$defs/cirrus-cs42l43-state" + - patternProperties: + "-pins$": + $ref: "#/$defs/cirrus-cs42l43-state" + additionalProperties: false + + spi: + type: object + $ref: /schemas/spi/spi-controller.yaml# + unevaluatedProperties: false + +$defs: + cirrus-cs42l43-state: + type: object + + allOf: + - $ref: /schemas/pinctrl/pincfg-node.yaml# + - $ref: /schemas/pinctrl/pinmux-node.yaml# + + oneOf: + - required: [ groups ] + - required: [ pins ] + + additionalProperties: false + + properties: + groups: + enum: [ gpio1, gpio2, gpio3, asp, pdmout2, pdmout1, i2c, spi ] + + pins: + enum: [ gpio1, gpio2, gpio3, + asp_dout, asp_fsync, asp_bclk, + pdmout2_clk, pdmout2_data, pdmout1_clk, pdmout1_data, + i2c_sda, i2c_scl, + spi_miso, spi_sck, spi_ssb ] + + function: + enum: [ gpio, spdif, irq, mic-shutter, spk-shutter ] + + drive-strength: + description: Set drive strength in mA + enum: [ 1, 2, 4, 8, 9, 10, 12, 16 ] + + input-debounce: + description: Set input debounce in uS + enum: [ 0, 85 ] + +required: + - compatible + - reg + - vdd-p-supply + - vdd-a-supply + - vdd-io-supply + - vdd-cp-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l43: codec@1a { + compatible = "cirrus,cs42l43"; + reg = <0x1a>; + + vdd-p-supply = <&vdd5v0>; + vdd-a-supply = <&vdd1v8>; + vdd-io-supply = <&vdd1v8>; + vdd-cp-supply = <&vdd1v8>; + vdd-amp-supply = <&vdd5v0>; + + reset-gpios = <&gpio 0>; + + interrupt-controller; + #interrupt-cells = <2>; + interrupt-parent = <&gpio>; + interrupts = <56 IRQ_TYPE_LEVEL_LOW>; + + #sound-dai-cells = <1>; + + clocks = <&clks 0>; + clock-names = "mclk"; + + cs42l43_pins: pinctrl { + gpio-controller; + #gpio-cells = <2>; + gpio-ranges = <&cs42l43_pins 0 0 3>; + + pinctrl-names = "default"; + pinctrl-0 = <&pinsettings>; + + pinsettings: default-state { + shutter-pins { + groups = "gpio3"; + function = "mic-shutter"; + }; + }; + }; + + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs-gpios = <&cs42l43_pins 1 0>; + + sensor@0 { + compatible = "bosch,bme680"; + reg = <0>; + spi-max-frequency = <1400000>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml new file mode 100644 index 000000000..f7bafbd4f --- /dev/null +++ b/Documentation/devicetree/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: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs42l51@4a { + compatible = "cirrus,cs42l51"; + reg = <0x4a>; + #sound-dai-cells = <0>; + clocks = <&mclk_prov>; + clock-names = "MCLK"; + VL-supply = <®_audio>; + VD-supply = <®_audio>; + VA-supply = <®_audio>; + VAHP-supply = <®_audio>; + reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; + + /* assume audio-graph */ + port { + cpu_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l84.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l84.yaml new file mode 100644 index 000000000..7f8338e8a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l84.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l84.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS42L84 audio CODEC + +maintainers: + - Martin PoviÅ¡er <povik+lin@cutebit.org> + +description: | + The CS42L84 is a headphone jack codec made by Cirrus Logic and embedded + in personal computers sold by Apple. It was first seen in 2021 Macbook + Pro models. It has stereo DAC for playback, mono ADC for capture, and + is somewhat similar to CS42L42 but with a different regmap. + +properties: + compatible: + enum: + - cirrus,cs42l84 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + jack_codec: codec@4b { + compatible = "cirrus,cs42l84"; + reg = <0x4b>; + reset-gpios = <&pinctrl_nub 4 GPIO_ACTIVE_LOW>; + interrupts-extended = <&pinctrl_ap 180 IRQ_TYPE_LEVEL_LOW>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml new file mode 100644 index 000000000..a1ae548c4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml @@ -0,0 +1,111 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42xx8.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS42448/CS42888 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +properties: + compatible: + enum: + - cirrus,cs42448 + - cirrus,cs42888 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 6000000 + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + const: mclk + + VA-supply: + description: Analog power supply. + + VD-supply: + description: Digital power supply. + + VLC-supply: + description: Control port power supply + + VLS-supply: + description: Serial port interface power supply. + + reset-gpios: + description: This pin is connected to the chip's RESET pin. + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - clocks + - clock-names + +if: + properties: + compatible: + contains: + const: cirrus,cs42888 +then: + required: + - VA-supply + - VD-supply + - VLC-supply + - VLS-supply + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <®_audio>; + VD-supply = <®_audio>; + VLS-supply = <®_audio>; + VLC-supply = <®_audio>; + reset-gpios = <&gpio 1>; + }; + }; + + spi { + #address-cells = <1>; + #size-cells = <0>; + cs-gpios = <&gpio 8 0>; + + codec@0 { + compatible = "cirrus,cs42888"; + reg = <0>; + spi-max-frequency = <6000000>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <®_audio>; + VD-supply = <®_audio>; + VLS-supply = <®_audio>; + VLC-supply = <®_audio>; + reset-gpios = <&gpio 1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs48l32.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs48l32.yaml new file mode 100644 index 000000000..bf087b57a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs48l32.yaml @@ -0,0 +1,195 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs48l32.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS48L32 audio DSP. + +maintainers: + - patches@opensource.cirrus.com + +description: | + The CS48L32 is a high-performance low-power audio DSP for smartphones and + other portable audio devices. The CS48L32 combines a programmable Halo Core + DSP with a variety of power-efficient fixed-function audio processors. + + See also the binding headers: + + include/dt-bindings/sound/cs48l32.yaml + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,cs48l32 + + reg: + description: SPI chip-select number. + maxItems: 1 + + spi-max-frequency: + maximum: 25000000 + + vdd-a-supply: + description: Regulator supplying VDD_A + + vdd-d-supply: + description: Regulator supplying VDD_D + + vdd-io-supply: + description: Regulator supplying VDD_IO + + vdd-cp-supply: + description: Regulator supplying VDD_CP + + reset-gpios: + description: + One entry specifying the GPIO controlling /RESET. Although optional, + it is strongly recommended to use a hardware reset. + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: The clock supplied on MCLK1 + + clock-names: + const: mclk1 + + '#sound-dai-cells': + const: 1 + + cirrus,in-type: + description: | + A list of input type settings for each ADC input. + Inputs are one of these types: + CS48L32_IN_TYPE_DIFF : analog differential (default) + CS48L32_IN_TYPE_SE : analog single-ended + + The type of the left (L) and right (R) channel on each input is + independently configured, as are the two groups of pins muxable to + the input (referred to in the datasheet as "1" and "2"). + + $ref: /schemas/types.yaml#/definitions/uint32-array + items: + - description: + IN1L_1 analog input type. One of the CS48L32_IN_TYPE_xxx. + minimum: 0 + maximum: 1 + default: 0 + - description: + IN1R_1 analog input type. One of the CS48L32_IN_TYPE_xxx. + minimum: 0 + maximum: 1 + default: 0 + - description: + IN1L_2 analog input type. One of the CS48L32_IN_TYPE_xxx. + minimum: 0 + maximum: 1 + default: 0 + - description: + IN1R_2 analog input type. One of the CS48L32_IN_TYPE_xxx. + minimum: 0 + maximum: 1 + default: 0 + + cirrus,pdm-sup: + description: | + Indicate which MICBIAS output supplies bias to the microphone. + There is one cell per input (IN1, IN2, ...). + + One of the CS48L32_MICBIAS_xxx values. + CS48L32_PDM_SUP_VOUT_MIC : mic biased from VOUT_MIC + CS48L32_PDM_SUP_MICBIAS1 : mic biased from MICBIAS1 + + Also see the INn_PDM_SUP field in the datasheet. + $ref: /schemas/types.yaml#/definitions/uint32-array + items: + - description: IN1 PDM supply source + minimum: 0 + maximum: 1 + default: 0 + - description: IN2 PDM supply source + minimum: 0 + maximum: 1 + default: 0 + +required: + - compatible + - reg + - vdd-a-supply + - vdd-d-supply + - vdd-io-supply + - vdd-cp-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/cs48l32.h> + + spi@e0006000 { + #address-cells = <1>; + #size-cells = <0>; + reg = <0xe0006000 0x1000>; + + codec@1 { + compatible = "cirrus,cs48l32"; + + reg = <0x1>; + spi-max-frequency = <2500000>; + + vdd-a-supply = <®ulator_1v8>; + vdd-d-supply = <®ulator_1v2>; + vdd-io-supply = <®ulator_1v8>; + vdd-cp-supply = <®ulator_1v8>; + + reset-gpios = <&gpio 0 0>; + + clocks = <&clks 0>; + clock-names = "mclk1"; + + interrupt-parent = <&gpio0>; + interrupts = <56 8>; + + #sound-dai-cells = <1>; + + cirrus,in-type = < + CS48L32_IN_TYPE_DIFF CS48L32_IN_TYPE_DIFF + CS48L32_IN_TYPE_SE CS48L32_IN_TYPE_SE + >; + + cirrus,pdm-sup = < + CS48L32_PDM_SUP_MICBIAS1 CS48L32_PDM_SUP_MICBIAS1 + >; + }; + }; + +# +# Minimal config +# + - | + #include <dt-bindings/sound/cs48l32.h> + + spi@e0006000 { + #address-cells = <1>; + #size-cells = <0>; + reg = <0xe0006000 0x1000>; + + codec@1 { + compatible = "cirrus,cs48l32"; + + reg = <0x1>; + + vdd-a-supply = <®ulator_1v8>; + vdd-d-supply = <®ulator_1v2>; + vdd-io-supply = <®ulator_1v8>; + vdd-cp-supply = <®ulator_1v8>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs530x.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs530x.yaml new file mode 100644 index 000000000..7600fff0e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs530x.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs530x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic cs530x family of audio ADCs + +maintainers: + - Paul Handrigan <paulha@opensource.cirrus.com> + - patches@opensource.cirrus.com + +description: + The CS530X devices are a family of high performance audio ADCs. + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +properties: + compatible: + enum: + - cirrus,cs4282 + - cirrus,cs4302 + - cirrus,cs4304 + - cirrus,cs4308 + - cirrus,cs5302 + - cirrus,cs5304 + - cirrus,cs5308 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + '#sound-dai-cells': + const: 1 + + reset-gpios: + maxItems: 1 + + vdd-a-supply: + description: Analog power supply + + vdd-io-supply: + description: Digital IO power supply + + cirrus,in-hiz-pin12: + description: + Sets input channels one and two to high impedance. + type: boolean + + cirrus,in-hiz-pin34: + description: + Sets input channels three and four to high impedance. + type: boolean + + cirrus,in-hiz-pin56: + description: + Sets input channels five and six to high impedance. + type: boolean + + cirrus,in-hiz-pin78: + description: + Sets input channels seven and eight to high impedance. + type: boolean + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + cs5304: adc@48 { + compatible = "cirrus,cs5304"; + reg = <0x48>; + #sound-dai-cells = <1>; + reset-gpios = <&gpio 110 GPIO_ACTIVE_LOW>; + vdd-a-supply = <&vreg>; + vdd-io-supply = <&vreg>; + cirrus,in-hiz-pin34; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml b/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml new file mode 100644 index 000000000..4693e85ae --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + dmas: + items: + - description: out DMA channel + - description: in DMA channel + + dma-names: + items: + - const: tx + - const: rx + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - '#sound-dai-cells' + - reg + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + i2s: i2s@80820000 { + compatible = "cirrus,ep9301-i2s"; + #sound-dai-cells = <0>; + reg = <0x80820000 0x100>; + interrupt-parent = <&vic1>; + interrupts = <28>; + clocks = <&syscon 29>, + <&syscon 30>, + <&syscon 31>; + clock-names = "mclk", "sclk", "lrclk"; + dmas = <&dma0 0 1>, <&dma0 0 2>; + dma-names = "tx", "rx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml new file mode 100644 index 000000000..52f024f53 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - cirrus,lochnagar2-soundcard + + '#sound-dai-cells': + description: + The first cell indicating the audio interface. + const: 1 + + clocks: + description: + Master clock source for the sound card, should normally be set to + LOCHNAGAR_SOUNDCARD_MCLK provided by the Lochnagar clock driver. + maxItems: 1 + + clock-names: + const: mclk + +required: + - compatible + - '#sound-dai-cells' + - clocks + - clock-names + +unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/cirrus,madera.yaml b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml new file mode 100644 index 000000000..014d4eaa8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,madera.yaml @@ -0,0 +1,118 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/cix,sky1-ipbloq-hda.yaml b/Documentation/devicetree/bindings/sound/cix,sky1-ipbloq-hda.yaml new file mode 100644 index 000000000..02ac5f1aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cix,sky1-ipbloq-hda.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cix,sky1-ipbloq-hda.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: CIX IPBLOQ HDA controller + +description: + CIX IPBLOQ High Definition Audio (HDA) Controller + +maintainers: + - Joakim Zhang <joakim.zhang@cixtech.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: cix,sky1-ipbloq-hda + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 2 + + clock-names: + items: + - const: ipg + - const: per + + resets: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - resets + +unevaluatedProperties: false + +examples: + - | + #include<dt-bindings/interrupt-controller/arm-gic.h> + + hda@70c0000 { + compatible = "cix,sky1-ipbloq-hda"; + reg = <0x70c0000 0x10000>; + interrupts = <GIC_SPI 234 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audss_clk 7>, + <&audss_clk 8>; + clock-names = "ipg", "per"; + resets = <&audss_rst 14>; + model = "CIX SKY1 EVB HDA"; + }; diff --git a/Documentation/devicetree/bindings/sound/component-common.yaml b/Documentation/devicetree/bindings/sound/component-common.yaml new file mode 100644 index 000000000..37766c5f3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/component-common.yaml @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/component-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Component Common Properties + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: | + Card implementing the routing property define the connection between + audio components as list of string pair. Component using the same + sink/source names may use this property to prepend the name of their + sinks/sources with the provided string. + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/cs35l32.txt b/Documentation/devicetree/bindings/sound/cs35l32.txt new file mode 100644 index 000000000..1417d3f5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l32.txt @@ -0,0 +1,62 @@ +CS35L32 audio CODEC + +Required properties: + + - compatible : "cirrus,cs35l32" + + - reg : the I2C address of the device for I2C. Address is determined by the level + of the AD0 pin. Level 0 is 0x40 while Level 1 is 0x41. + + - VA-supply, VP-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - cirrus,boost-manager : Boost voltage control. + 0 = Automatically managed. Boost-converter output voltage is the higher + of the two: Class G or adaptive LED voltage. + 1 = Automatically managed irrespective of audio, adapting for low-power + dissipation when LEDs are ON, and operating in Fixed-Boost Bypass Mode + if LEDs are OFF (VBST = VP). + 2 = (Default) Boost voltage fixed in Bypass Mode (VBST = VP). + 3 = Boost voltage fixed at 5 V. + + - cirrus,sdout-datacfg : Data configuration for dual CS35L32 applications only. + Determines the data packed in a two-CS35L32 configuration. + 0 = Left/right channels VMON[11:0], IMON[11:0], VPMON[7:0]. + 1 = Left/right channels VMON[11:0], IMON[11:0], STATUS. + 2 = (Default) left/right channels VMON[15:0], IMON [15:0]. + 3 = Left/right channels VPMON[7:0], STATUS. + + - cirrus,sdout-share : SDOUT sharing. Determines whether one or two CS35L32 + devices are on board sharing SDOUT. + 0 = (Default) One IC. + 1 = Two IC's. + + - cirrus,battery-recovery : Low battery nominal recovery threshold, rising VP. + 0 = 3.1V + 1 = 3.2V + 2 = 3.3V (Default) + 3 = 3.4V + + - cirrus,battery-threshold : Low battery nominal threshold, falling VP. + 0 = 3.1V + 1 = 3.2V + 2 = 3.3V + 3 = 3.4V (Default) + 4 = 3.5V + 5 = 3.6V + +Example: + +codec: codec@40 { + compatible = "cirrus,cs35l32"; + reg = <0x40>; + reset-gpios = <&gpio 10 0>; + cirrus,boost-manager = <0x03>; + cirrus,sdout-datacfg = <0x02>; + VA-supply = <®_audio>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l33.txt b/Documentation/devicetree/bindings/sound/cs35l33.txt new file mode 100644 index 000000000..dc5a355d1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l33.txt @@ -0,0 +1,124 @@ +CS35L33 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l33" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : gpio used to reset the amplifier + + - interrupts : IRQ line info CS35L33. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-ctl : Booster voltage use to supply the amp. If the value is + 0, then VBST = VP. If greater than 0, the boost voltage will be 3300mV with + a value of 1 and will increase at a step size of 100mV until a maximum of + 8000mV. + + - cirrus,ramp-rate : On power up, it affects the time from when the power + up sequence begins to the time the audio reaches a full-scale output. + On power down, it affects the time from when the power-down sequence + begins to when the amplifier disables the PWM outputs. If this property + is not set then soft ramping will be disabled and ramp time would be + 20ms. If this property is set to 0,1,2,3 then ramp times would be 40ms, + 60ms,100ms,175ms respectively for 48KHz sample rate. + + - cirrus,boost-ipk : The maximum current allowed for the boost converter. + The range starts at 1850000uA and goes to a maximum of 3600000uA + with a step size of 15625uA. The default is 2500000uA. + + - cirrus,imon-adc-scale : Configures the scaling of data bits from the IMON + ADC data word. This property can be set as a value of 0 for bits 15 down + to 0, 6 for 21 down to 6, 7, for 22 down to 7, 8 for 23 down to 8. + + +Optional H/G Algorithm sub-node: + +The cs35l33 node can have a single "cirrus,hg-algo" sub-node that will enable +the internal H/G Algorithm. + + - cirrus,hg-algo : Sub-node for internal Class H/G algorithm that + controls the amplifier supplies. + +Optional properties for the "cirrus,hg-algo" sub-node: + + - cirrus,mem-depth : Memory depth for the Class H/G algorithm measured in + LRCLK cycles. If this property is set to 0, 1, 2, or 3 then the memory + depths will be 1, 4, 8, 16 LRCLK cycles. The default is 16 LRCLK cycles. + + cirrus,release-rate : The number of consecutive LRCLK periods before + allowing release condition tracking updates. The number of LRCLK periods + start at 3 to a maximum of 255. + + - cirrus,ldo-thld : Configures the signal threshold at which the PWM output + stage enters LDO operation. Starts as a default value of 50mV for a value + of 1 and increases with a step size of 50mV to a maximum of 750mV (value of + 0xF). + + - cirrus,ldo-path-disable : This is a boolean property. If present, the H/G + algorithm uses the max detection path. If not present, the LDO + detection path is used. + + - cirrus,ldo-entry-delay : The LDO entry delay in milliseconds before the H/G + algorithm switches to the LDO voltage. This property can be set to values + from 0 to 7 for delays of 5ms, 10ms, 50ms, 100ms, 200ms, 500ms, 1000ms. + The default is 100ms. + + - cirrus,vp-hg-auto : This is a boolean property. When set, class H/G VPhg + automatic updating is enabled. + + - cirrus,vp-hg : Class H/G algorithm VPhg. Controls the H/G algorithm's + reference to the VP voltage for when to start generating a boosted VBST. + The reference voltage starts at 3000mV with a value of 0x3 and is increased + by 100mV per step to a maximum of 5500mV. + + - cirrus,vp-hg-rate : The rate (number of LRCLK periods) at which the VPhg is + allowed to increase to a higher voltage when using VPhg automatic + tracking. This property can be set to values from 0 to 3 with rates of 128 + periods, 2048 periods, 32768 periods, and 524288 periods. + The default is 32768 periods. + + - cirrus,vp-hg-va : VA calculation reference for automatic VPhg tracking + using VPMON. This property can be set to values from 0 to 6 starting at + 1800mV with a step size of 50mV up to a maximum value of 1750mV. + Default is 1800mV. + +Example: + +cs35l33: cs35l33@40 { + compatible = "cirrus,cs35l33"; + reg = <0x40>; + + VA-supply = <&ldo5_reg>; + VP-supply = <&ldo5_reg>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&cs47l91 34 0>; + + cirrus,ramp-rate = <0x0>; + cirrus,boost-ctl = <0x30>; /* VBST = 8000mV */ + cirrus,boost-ipk = <0xE0>; /* 3600mA */ + cirrus,imon-adc-scale = <0> /* Bits 15 down to 0 */ + + cirrus,hg-algo { + cirrus,mem-depth = <0x3>; + cirrus,release-rate = <0x3>; + cirrus,ldo-thld = <0x1>; + cirrus,ldo-path-disable = <0x0>; + cirrus,ldo-entry-delay=<0x4>; + cirrus,vp-hg-auto; + cirrus,vp-hg=<0xF>; + cirrus,vp-hg-rate=<0x2>; + cirrus,vp-hg-va=<0x0>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l34.txt b/Documentation/devicetree/bindings/sound/cs35l34.txt new file mode 100644 index 000000000..2f7606b7d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l34.txt @@ -0,0 +1,62 @@ +CS35L34 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l34" + + - reg : the I2C address of the device for I2C. + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-vtge-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from VP to 8V with + increments of 100mV. + + - cirrus,boost-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + + - reset-gpios: GPIO used to reset the amplifier. + + - interrupts : IRQ line info CS35L34. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The + range starts at 1200mA and goes to a maximum of 3840mA with increments of + 80mA. The default value is 2480mA. + + - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the + received mono data is in the left or right portion of the I2S frame + according to the AD0 pin or directly via this configuration. + 0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel), + 0x2 = Left, + 0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right + channel), + 0x3 = Right. + + - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take + effect without waiting for a zero cross. + + - cirrus,tdm-rising-edge: Boolean property. If set, data is on the rising edge of + SCLK. Otherwise, data is on the falling edge of SCLK. + + +Example: + +cs35l34: cs35l34@40 { + compatible = "cirrus,cs35l34"; + reg = <0x40>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&gpio 10 0>; + + cirrus,boost-vtge-milltvolt = <8000>; /* 8V */ + cirrus,boost-ind-nanohenry = <1000>; /* 1uH */ + cirrus,boost-peak-milliamp = <3000>; /* 3A */ +}; diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 000000000..e84f30c5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,181 @@ +CS35L35 Boosted Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-ind-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + - reset-gpios : gpio used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,external-boost : Boolean to specify the device is using an external + boost supply, note that sharing a boost from another cs35l35 would constitute + using an external supply for the slave device + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + - cirrus,sp-drv-unused : Determines how unused slots should be driven on the + Serial Port. + 0 - Hi-Z + 2 - Drive 0's (Default) + 3 - Drive 1's + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2600mV to 9000mV with + increments of 100mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1680mA and goes to a maximum of 4480mA with increments of + 110mA. + (Default) 2.46 Amps + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling 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/Documentation/devicetree/bindings/sound/cs42l52.txt b/Documentation/devicetree/bindings/sound/cs42l52.txt new file mode 100644 index 000000000..bc03c9312 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l52.txt @@ -0,0 +1,46 @@ +CS42L52 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l52" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - cirrus,reset-gpio : GPIO controller's phandle and the number + of the GPIO used to reset the codec. + + - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency. + Allowable values of 0x00 through 0x0F. These are raw values written to the + register, not the actual frequency. The frequency is determined by the following. + Frequency = (64xFs)/(N+2) + N = chgfreq_val + Fs = Sample Rate (variable) + + - cirrus,mica-differential-cfg : boolean, If present, then the MICA input is configured + as a differential input. If not present then the MICA input is configured as + Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input. + + - cirrus,micb-differential-cfg : boolean, If present, then the MICB input is configured + as a differential input. If not present then the MICB input is configured as + Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input. + + - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin + 0 = 0.5 x VA + 1 = 0.6 x VA + 2 = 0.7 x VA + 3 = 0.8 x VA + 4 = 0.83 x VA + 5 = 0.91 x VA + +Example: + +codec: codec@4a { + compatible = "cirrus,cs42l52"; + reg = <0x4a>; + reset-gpio = <&gpio 10 0>; + cirrus,chgfreq-divisor = <0x05>; + cirrus.mica-differential-cfg; + cirrus,micbias-lvl = <5>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs42l56.txt b/Documentation/devicetree/bindings/sound/cs42l56.txt new file mode 100644 index 000000000..4ba520a28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l56.txt @@ -0,0 +1,63 @@ +CS42L52 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l56" + + - reg : the I2C address of the device for I2C + + - VA-supply, VCP-supply, VLDO-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - cirrus,gpio-nreset : GPIO controller's phandle and the number + of the GPIO used to reset the codec. + + - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency. + Allowable values of 0x00 through 0x0F. These are raw values written to the + register, not the actual frequency. The frequency is determined by the following. + Frequency = MCLK / 4 * (N+2) + N = chgfreq_val + MCLK = Where MCLK is the frequency of the mclk signal after the MCLKDIV2 circuit. + + - cirrus,ain1a-ref-cfg, ain1b-ref-cfg : boolean, If present, AIN1A or AIN1B are configured + as a pseudo-differential input referenced to AIN1REF/AIN3A. + + - cirrus,ain2a-ref-cfg, ain2b-ref-cfg : boolean, If present, AIN2A or AIN2B are configured + as a pseudo-differential input referenced to AIN2REF/AIN3B. + + - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin. + 0 = 0.5 x VA + 1 = 0.6 x VA + 2 = 0.7 x VA + 3 = 0.8 x VA + 4 = 0.83 x VA + 5 = 0.91 x VA + + - cirrus,adaptive-pwr-cfg : Configures how the power to the Headphone and Lineout + Amplifiers adapt to the output signal levels. + 0 = Adapt to Volume Mode. Voltage level determined by the sum of the relevant volume settings. + 1 = Fixed - Headphone and Line Amp supply = + or - VCP/2. + 2 = Fixed - Headphone and Line Amp supply = + or - VCP. + 3 = Adapted to Signal; Voltage level is dynamically determined by the output signal. + + - cirrus,hpf-left-freq, hpf-right-freq : Sets the corner frequency (-3dB point) for the internal High-Pass + Filter. + 0 = 1.8Hz + 1 = 119Hz + 2 = 236Hz + 3 = 464Hz + + +Example: + +codec: codec@4b { + compatible = "cirrus,cs42l56"; + reg = <0x4b>; + cirrus,gpio-nreset = <&gpio 10 0>; + cirrus,chgfreq-divisor = <0x05>; + cirrus.ain1_ref_cfg; + cirrus,micbias-lvl = <5>; + VA-supply = <®_audio>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt new file mode 100644 index 000000000..47b868b5a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l73.txt @@ -0,0 +1,22 @@ +CS42L73 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l73" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset_gpio : a GPIO spec for the reset pin. + - chgfreq : Charge Pump Frequency values 0x00-0x0F + + +Example: + +codec: cs42l73@4a { + compatible = "cirrus,cs42l73"; + reg = <0x4a>; + reset_gpio = <&gpio 10 0>; + chgfreq = <0x05>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 000000000..8b1dd5aeb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,67 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130", "cirrus,cs4399", "cirrus,cs43131", + "cirrus,cs43198" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + + - cirrus,dc-measure: + Boolean, define to enable headphone DC impedance measurement. + + - cirrus,ac-measure: + Boolean, define to enable headphone AC impedance measurement. + DC impedance must also be enabled for AC impedance measurement. + + - cirrus,dc-threshold: + Define 2 DC impedance thresholds in ohms for HP output control. + Default values are 50 and 120 Ohms. + + - cirrus,ac-freq: + Define the frequencies at which to measure HP AC impedance. + Only used if "cirrus,dc-measure" is defined. + Exactly 10 frequencies must be defined. + If this properties is undefined, by default, + following frequencies are used: + <24 43 93 200 431 928 2000 4309 9283 20000> + The above frequencies are logarithmically equally spaced. + Log base is 10. + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + cirrus,dc-measure; + cirrus,ac-measure; + cirrus,dc-threshold = /bits/ 16 <20 100>; + cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 000000000..dc256adb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,44 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + + - VA-supply, VP-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either + GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled + by the driver automatically. + + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. + 0 = Hi-Z + 1 = 1.80 V + 2 = 2.75 V + + - cirrus,use-sdout2 : This is a boolean property. If present, it indicates + the hardware design connects both SDOUT1 and SDOUT2 + pins to output data. Otherwise, it indicates that + only SDOUT1 is connected for data output. + * CS53l30 supports 4-channel data output in the same + * frame using two different ways: + * 1) Normal I2S mode on two data pins -- each SDOUT + * carries 2-channel data in the same time. + * 2) TDM mode on one 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/Documentation/devicetree/bindings/sound/da7218.txt b/Documentation/devicetree/bindings/sound/da7218.txt new file mode 100644 index 000000000..2cf30899b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da7218.txt @@ -0,0 +1,102 @@ +Dialog Semiconductor DA7218 Audio Codec bindings + +DA7218 is an audio codec with HP detect feature. + +====== + +Required properties: +- compatible : Should be "dlg,da7217" or "dlg,da7218" +- reg: Specifies the I2C slave address + +- VDD-supply: VDD power supply for the device +- VDDMIC-supply: VDDMIC power supply for the device +- VDDIO-supply: VDDIO power supply for the device + (See Documentation/devicetree/bindings/regulator/regulator.txt for further + information relating to regulators) + +Optional properties: +- interrupts: IRQ line info for DA7218 chip. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for + further information relating to interrupt properties) +- interrupt-names : Name associated with interrupt line. Should be "wakeup" if + interrupt is to be used to wake system, otherwise "irq" should be used. +- wakeup-source: Flag to indicate this device can wake system (suspend/resume). + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias1-lvl-millivolt : Voltage (mV) for Mic Bias 1 + [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>] +- dlg,micbias2-lvl-millivolt : Voltage (mV) for Mic Bias 2 + [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>] +- dlg,mic1-amp-in-sel : Mic1 input source type + ["diff", "se_p", "se_n"] +- dlg,mic2-amp-in-sel : Mic2 input source type + ["diff", "se_p", "se_n"] +- dlg,dmic1-data-sel : DMIC1 channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic1-samplephase : When to sample audio from DMIC1. + ["on_clkedge", "between_clkedge"] +- dlg,dmic1-clkrate-hz : DMic1 clock frequency (Hz). + [<1500000>, <3000000>] +- dlg,dmic2-data-sel : DMic2 channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic2-samplephase : When to sample audio from DMic2. + ["on_clkedge", "between_clkedge"] +- dlg,dmic2-clkrate-hz : DMic2 clock frequency (Hz). + [<1500000>, <3000000>] +- dlg,hp-diff-single-supply : Boolean flag, use single supply for HP + (DA7217 only) + +====== + +Optional Child node - 'da7218_hpldet' (DA7218 only): + +Optional properties: +- dlg,jack-rate-us : Time between jack detect measurements (us) + [<5>, <10>, <20>, <40>, <80>, <160>, <320>, <640>] +- dlg,jack-debounce : Number of debounce measurements taken for jack detect + [<0>, <2>, <3>, <4>] +- dlg,jack-threshold-pct : Threshold level for jack detection (% of VDD) + [<84>, <88>, <92>, <96>] +- dlg,comp-inv : Boolean flag, invert comparator output +- dlg,hyst : Boolean flag, enable hysteresis +- dlg,discharge : Boolean flag, auto discharge of Mic Bias on jack removal + +====== + +Example: + + codec: da7218@1a { + compatible = "dlg,da7218"; + reg = <0x1a>; + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + wakeup-source; + + VDD-supply = <®_audio>; + VDDMIC-supply = <®_audio>; + VDDIO-supply = <®_audio>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias1-lvl-millivolt = <2600>; + dlg,micbias2-lvl-millivolt = <2600>; + dlg,mic1-amp-in-sel = "diff"; + dlg,mic2-amp-in-sel = "diff"; + + dlg,dmic1-data-sel = "lrise_rfall"; + dlg,dmic1-samplephase = "on_clkedge"; + dlg,dmic1-clkrate-hz = <3000000>; + dlg,dmic2-data-sel = "lrise_rfall"; + dlg,dmic2-samplephase = "on_clkedge"; + dlg,dmic2-clkrate-hz = <3000000>; + + da7218_hpldet { + dlg,jack-rate-us = <40>; + dlg,jack-debounce = <2>; + dlg,jack-threshold-pct = <84>; + dlg,hyst; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/dai-common.yaml b/Documentation/devicetree/bindings/sound/dai-common.yaml new file mode 100644 index 000000000..1aed2f0f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dai-common.yaml @@ -0,0 +1,18 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dai-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface Common Properties + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: component-common.yaml# + +properties: + '#sound-dai-cells': true + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/dai-params.yaml b/Documentation/devicetree/bindings/sound/dai-params.yaml new file mode 100644 index 000000000..cd8508175 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dai-params.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dai-params.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface (DAI) Stream Parameters + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +select: false + +properties: + convert-channels: + description: Number of audio channels used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 32 + + convert-sample-format: + description: Audio sample format used by DAI + $ref: /schemas/types.yaml#/definitions/string + enum: + - s8 + - s16_le + - s24_le + - s24_3le + - s32_le + + convert-rate: + description: Audio sample rate used by DAI + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 8000 + maximum: 192000 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 000000000..963e10051 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,49 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx +- ti,model : The user-visible name of this sound complex. +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,mcasp-controller : The phandle of the McASP controller +- ti,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the codec's pins, and the jacks on the board: + +Optional properties: +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec. +- clocks : Reference to the master clock +- clock-names : The clock should be named "mclk" +- Either codec-clock-rate or the codec-clock reference has to be defined. If + the both are defined the driver attempts to set referenced clock to the + defined rate and takes the rate from the clock reference. + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + * Line In + + +Example: + +sound { + compatible = "ti,da830-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + ti,audio-routing = + "Headphone Jack", "HPLOUT", + "Headphone Jack", "HPROUT", + "Line Out", "LLOUT", + "Line Out", "RLOUT", + "MIC3L", "Mic Bias 2V", + "MIC3R", "Mic Bias 2V", + "Mic Bias 2V", "Mic Jack", + "LINE1L", "Line In", + "LINE2L", "Line In", + "LINE1R", "Line In", + "LINE2R", "Line In"; +}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000..f8a602cee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,263 @@ +# 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. This property + specifies the TX playback TDM slot count, along with default RX slot count + if tdm-slots-rx is not specified. + The property is ignored in DIT mode. + minimum: 2 + maximum: 32 + + tdm-slots-rx: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of RX capture channels over one serializer. If specified, + allows independent RX TDM slot count separate from TX. Requires + ti,async-mode to be enabled for independent TX/RX clock rates. + The property is ignored in DIT mode. + minimum: 2 + maximum: 32 + + ti,async-mode: + description: + Specify to allow independent TX & RX clocking, + to enable audio playback & record with different sampling rate, + and different number of bits per frame. + if property is omitted, TX and RX will share same bit clock and frame clock signals, + thus RX need to use same bits per frame and sampling rate as TX in synchronous mode. + the property is ignored in DIT mode (as DIT is TX-only) + type: boolean + + 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: + minItems: 1 + maxItems: 2 + + interrupt-names: + oneOf: + - description: TX interrupt + const: tx + - description: RX interrupt + const: rx + - description: TX and RX interrupts + items: + - const: tx + - const: rx + - description: Common/combined interrupt + 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. This property specifies + the TX ratio, along with default RX ratio if auxclk-fs-ratio-rx + is not specified. + When not specified, the inputted system clock frequency via set_sysclk + callback by the machine driver is used for divider calculation. + + auxclk-fs-ratio-rx: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Ratio of AUCLK and FS rate for RX. If specified, allows + for a different RX ratio. Requires ti,async-mode to be + enabled when the ratio differs from auxclk-fs-ratio. + When not specified, it defaults to the value of auxclk-fs-ratio. + The property is ignored in DIT mode. + + 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: + - $ref: dai-common.yaml# + - if: + properties: + op-mode: + enum: + - 0 + then: + required: + - tdm-slots + + - if: + properties: + op-mode: + const: 1 + then: + properties: + tdm-slots: false + tdm-slots-rx: false + ti,async-mode: false + auxclk-fs-ratio-rx: false + + - if: + required: + - tdm-slots-rx + then: + required: + - ti,async-mode + + - if: + required: + - auxclk-fs-ratio-rx + then: + required: + - ti,async-mode + +unevaluatedProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + ti,async-mode; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml new file mode 100644 index 000000000..4fa677023 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml @@ -0,0 +1,113 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McBSP Controller for TI SoCs + +maintainers: + - Bastien Curutchet <bastien.curutchet@bootlin.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,da850-mcbsp + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + dmas: + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + items: + - const: tx + - const: rx + + interrupts: + items: + - description: RX interrupt + - description: TX interrupt + + interrupt-names: + items: + - const: rx + - const: tx + + clocks: + minItems: 1 + items: + - description: functional clock + - description: external input clock for sample rate generator. + + clock-names: + minItems: 1 + items: + - const: fck + - const: clks + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + ti,T1-framing-tx: + $ref: /schemas/types.yaml#/definitions/flag + description: + If the property is present, tx data delay is set to 2 bit clock periods. + McBSP will insert a blank period (high-impedance period) before the first + data bit. This can be used to interface to T1-framing devices. + + ti,T1-framing-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: + If the property is present, rx data delay is set to 2 bit clock periods. + McBSP will discard the bit preceding the data stream (called framing bit). + This can be used to interface to T1-framing devices. + +required: + - "#sound-dai-cells" + - compatible + - reg + - reg-names + - dmas + - dma-names + - clocks + +unevaluatedProperties: false + +examples: + - | + mcbsp0@1d10000 { + #sound-dai-cells = <0>; + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x111000 0x1000>, + <0x311000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97>, <98>; + interrupt-names = "rx", "tx"; + dmas = <&edma0 3 1>, + <&edma0 2 1>; + dma-names = "tx", "rx"; + + clocks = <&psc1 14>; + }; diff --git a/Documentation/devicetree/bindings/sound/dialog,da7219.yaml b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml new file mode 100644 index 000000000..19137abdb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml @@ -0,0 +1,238 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7219 Audio Codec + +maintainers: + - David Rau <David.Rau.opensource@dm.renesas.com> + +description: + The DA7219 is an ultra low-power audio codec with + in-built advanced accessory detection (AAD) for mobile + computing and accessory applications, which supports + sample rates up to 96 kHz at 24-bit resolution. + +properties: + compatible: + const: dlg,da7219 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + VDD-supply: + description: + VDD power supply for the device. + + VDDMIC-supply: + description: + VDDMIC power supply for the device. + + VDDIO-supply: + description: + VDDIO power supply for the device. + + interrupt-names: + description: + Should be "wakeup" if interrupt is to be used to wake system, + otherwise "irq" should be used. + enum: + - wakeup + - irq + + wakeup-source: + type: boolean + description: + Flag to indicate this device can wake system (suspend/resume). + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: + Name given for DAI WCLK and BCLK outputs. + + clocks: + maxItems: 1 + description: + phandle and clock specifier for codec MCLK. + + clock-names: + const: mclk + + dlg,micbias-lvl: + enum: [1600, 1800, 2000, 2200, 2400, 2600] + description: + Voltage (mV) for Mic Bias. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-amp-in-sel: + enum: [diff, se_p, se_n] + description: + Mic input source type. + + diff - Differential. + + se_p - MIC_P. + Positive differential analog microphone input. + + se_n - MIC_N. + Negative differential analog microphone input. + $ref: /schemas/types.yaml#/definitions/string + + da7219_aad: + type: object + additionalProperties: false + description: + Configuration of advanced accessory detection. + properties: + dlg,micbias-pulse-lvl: + enum: [2800, 2900] + description: + Mic bias higher voltage pulse level (mV). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,micbias-pulse-time: + description: + Mic bias higher voltage pulse duration (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + + dlg,btn-cfg: + enum: [2, 5, 10, 50, 100, 200, 500] + description: + Periodic button press measurements for 4-pole jack (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-det-thr: + enum: [200, 500, 750, 1000] + description: + Impedance threshold for mic detection measurement (Ohms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-deb: + enum: [5, 10, 20, 50, 100, 200, 500, 1000] + description: + Debounce time for jack insertion (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-det-pty: + enum: [low, high] + description: + Polarity for jack insertion detection. + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-det-rate: + enum: ["32_64", "64_128", "128_256", "256_512"] + description: + Jack type (3/4 pole) detection latency (ms). + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-rem-deb: + enum: [1, 5, 10, 20] + description: + Debounce time for jack removal (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,a-d-btn-thr: + description: + Impedance threshold between buttons A and D. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,d-b-btn-thr: + description: + Impedance threshold between buttons D and B. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,b-c-btn-thr: + description: + Impedance threshold between buttons B and C. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,c-mic-btn-thr: + description: + Impedance threshold between button C and Mic. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,btn-avg: + enum: [1, 2, 4, 8] + description: + Number of 8-bit readings for averaged button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,adc-1bit-rpt: + enum: [1, 2, 4, 8] + description: + Repeat count for 1-bit button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - interrupts + - VDD-supply + - VDDMIC-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <&vdd_reg>; + VDDMIC-supply = <&vddmic_reg>; + VDDIO-supply = <&vddio_reg>; + + #clock-cells = <1>; + clock-output-names = "da7219-dai-wclk", "da7219-dai-bclk"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-ins-det-pty = "low"; + dlg,jack-det-rate = "32_64"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/dlg,da7213.yaml b/Documentation/devicetree/bindings/sound/dlg,da7213.yaml new file mode 100644 index 000000000..c2dede1e8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dlg,da7213.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dlg,da7213.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7212/DA7213 Audio Codec + +maintainers: + - Support Opensource <support.opensource@diasemi.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - dlg,da7212 + - dlg,da7213 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + "#sound-dai-cells": + const: 0 + + dlg,micbias1-lvl: + description: Voltage (mV) for Mic Bias 1 + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1600, 2200, 2500, 3000 ] + + dlg,micbias2-lvl: + description: Voltage (mV) for Mic Bias 2 + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1600, 2200, 2500, 3000 ] + + dlg,dmic-data-sel: + description: DMIC channel select based on clock edge + enum: [ lrise_rfall, lfall_rrise ] + + dlg,dmic-samplephase: + description: When to sample audio from DMIC + enum: [ on_clkedge, between_clkedge ] + + dlg,dmic-clkrate: + description: DMIC clock frequency (Hz) + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1500000, 3000000 ] + + VDDA-supply: + description: Analogue power supply + + VDDIO-supply: + description: I/O power supply + + VDDMIC-supply: + description: Mic Bias + + VDDSP-supply: + description: Speaker supply + + ports: + $ref: audio-graph-port.yaml#/definitions/ports + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "dlg,da7213"; + reg = <0x1a>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + #sound-dai-cells = <0>; + + dlg,micbias1-lvl = <2500>; + dlg,micbias2-lvl = <2500>; + + dlg,dmic-data-sel = "lrise_rfall"; + dlg,dmic-samplephase = "between_clkedge"; + dlg,dmic-clkrate = <3000000>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/dmic-codec.yaml b/Documentation/devicetree/bindings/sound/dmic-codec.yaml new file mode 100644 index 000000000..83c23e029 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dmic-codec.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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 + + vref-supply: + description: Phandle to the digital microphone reference supply + + 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 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + dmic { + compatible = "dmic-codec"; + dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; + num-channels = <1>; + wakeup-delay-ms = <50>; + modeswitch-delay-ms = <35>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/eukrea,asoc-tlv320.yaml b/Documentation/devicetree/bindings/sound/eukrea,asoc-tlv320.yaml new file mode 100644 index 000000000..24b793feb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea,asoc-tlv320.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/eukrea,asoc-tlv320.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio complex for Eukrea boards with tlv320aic23 codec. + +maintainers: + - Frank Li <Frank.Li@nxp.com> + +properties: + compatible: + const: eukrea,asoc-tlv320 + + eukrea,model: + $ref: /schemas/types.yaml#/definitions/string + description: + The user-visible name of this sound complex. + + ssi-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the SSI controller. + + fsl,mux-int-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + The internal port of the i.MX audio muxer (AUDMUX). + Note: The AUDMUX port numbering should start at 1, which is consistent with + hardware manual. + minimum: 1 + maximum: 8 + + fsl,mux-ext-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + The external port of the i.MX audio muxer. + Note: The AUDMUX port numbering should start at 1, which is consistent with + hardware manual. + minimum: 1 + maximum: 8 + +required: + - compatible + - eukrea,model + - ssi-controller + - fsl,mux-int-port + - fsl,mux-ext-port + +additionalProperties: false + +examples: + - | + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es71x4.yaml b/Documentation/devicetree/bindings/sound/everest,es71x4.yaml new file mode 100644 index 000000000..efe9f3fd3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es71x4.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es71x4.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES7134/7144/7154 2 channels I2S analog to digital converter + +maintainers: + - Neil Armstrong <neil.armstrong@linaro.org> + +properties: + compatible: + enum: + - everest,es7134 + - everest,es7144 + - everest,es7154 + + VDD-supply: true + PVDD-supply: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - VDD-supply + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - everest,es7134 + - everest,es7144 + then: + properties: + PVDD-supply: false + + - if: + properties: + compatible: + contains: + enum: + - everest,es7154 + then: + required: + - PVDD-supply + +unevaluatedProperties: false + +examples: + - | + codec { + compatible = "everest,es7134"; + #sound-dai-cells = <0>; + VDD-supply = <&vdd_supply>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/everest,es7241.yaml b/Documentation/devicetree/bindings/sound/everest,es7241.yaml new file mode 100644 index 000000000..e5cfb40f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es7241.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es7241.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES7241 2 channels I2S analog to digital converter + +maintainers: + - Neil Armstrong <neil.armstrong@linaro.org> + +properties: + compatible: + enum: + - everest,es7241 + + reset-gpios: + maxItems: 1 + description: GPIO connected to the reset pin + + m0-gpios: + maxItems: 1 + description: GPIO connected to the m0 pin + + m1-gpios: + maxItems: 1 + description: GPIO connected to the m0 pin + + everest,sdout-pull-down: + type: boolean + description: + 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 + + VDDP-supply: true + VDDA-supply: true + VDDD-supply: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - VDDP-supply + - VDDA-supply + - VDDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + codec { + compatible = "everest,es7241"; + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + VDDP-supply = <&vddp_supply>; + VDDA-supply = <&vdda_supply>; + VDDD-supply = <&vddd_supply>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000..f4ff23120 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,110 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8311, ES8316 and ES8323 audio CODECs + +maintainers: + - Daniel Drake <drake@endlessm.com> + - Katsuhiro Suzuki <katsuhiro@katsuster.net> + - Matteo Martelli <matteomartelli3@gmail.com> + - Binbin Zhou <zhoubinbin@loongson.cn> + +description: | + Everest ES8311, ES8316 and ES8323 audio CODECs + + Pins on the device (for linking into audio routes): + + Outputs: + * LOUT: Left Analog Output + * ROUT: Right Analog Output + * MICBIAS: Microphone Bias + + Inputs: + * MIC1P: Microphone 1 Positive Analog Input + * MIC1N: Microphone 1 Negative Analog Input + * MIC2P: Microphone 2 Positive Analog Input + * MIC2N: Microphone 2 Negative Analog Input + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + not: + contains: + const: everest,es8316 + then: + properties: + avdd-supply: false + cpvdd-supply: false + dvdd-supply: false + pvdd-supply: false + +properties: + compatible: + enum: + - everest,es8311 + - everest,es8316 + - everest,es8323 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + avdd-supply: + description: Regulator providing the analog supply, from 2.0 V to 3.6 V + + cpvdd-supply: + description: Regulator providing the charge pump supply, from 1.6 V to 2.0 V + + dvdd-supply: + description: Regulator providing the digital core supply, from 1.6 V to 3.6 V + + pvdd-supply: + description: Regulator providing the digital I/O supply, from 1.6 V to 3.6 V + + interrupts: + maxItems: 1 + description: Headphone detect interrupt + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + avdd-supply = <®_3p3v>; + cpvdd-supply = <®_1p8v>; + dvdd-supply = <®_1p8v>; + pvdd-supply = <®_1p8v>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100644 index 000000000..b5594a9d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + interrupts: + maxItems: 1 + description: interrupt output for headset detection + + "#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 property to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x00 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [00]; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8328.yaml b/Documentation/devicetree/bindings/sound/everest,es8328.yaml new file mode 100644 index 000000000..ddddd7b14 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8328.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8328.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8328 audio CODEC + +description: + Everest Audio Codec, which can be connected via I2C or SPI. + Pins on the device (for linking into audio routes) are + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * LINPUT1 + * RINPUT1 + * LINPUT2 + * RINPUT2 + * Mic Bias + +maintainers: + - David Yang <yangxiaohua@everest-semi.com> + +properties: + compatible: + oneOf: + - enum: + - everest,es8328 + - items: + - enum: + - everest,es8388 + - const: everest,es8328 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: A 22.5792 or 11.2896 MHz clock + + DVDD-supply: + description: Regulator providing digital core supply voltage 1.8 - 3.6V + + AVDD-supply: + description: Regulator providing analog supply voltage 3.3V + + PVDD-supply: + description: Regulator providing digital IO supply voltage 1.8 - 3.6V + + HPVDD-supply: + description: Regulator providing analog output voltage 3.3V + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - clocks + - DVDD-supply + - AVDD-supply + - PVDD-supply + - HPVDD-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8328: codec@11 { + compatible = "everest,es8328"; + reg = <0x11>; + AVDD-supply = <®_3p3v>; + DVDD-supply = <®_3p3v>; + HPVDD-supply = <®_3p3v>; + PVDD-supply = <®_3p3v>; + clocks = <&clks 169>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8375.yaml b/Documentation/devicetree/bindings/sound/everest,es8375.yaml new file mode 100644 index 000000000..4a3d671c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8375.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8375 audio CODEC + +maintainers: + - Michael Zhang <zhangyi@everest-semi.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: everest,es8375 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + vdda-supply: + description: + Analogue power supply. + + vddd-supply: + description: + Interface power supply. + + everest,mclk-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + Represents the MCLK/SCLK pair pins used as the internal clock. + 0 represents selecting MCLK. + 1 represents selecting SCLK. + enum: [0, 1] + default: 0 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + - vdda-supply + - vddd-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8375: codec@18 { + compatible = "everest,es8375"; + reg = <0x18>; + vdda-supply = <&vdd3v3>; + vddd-supply = <&vdd3v3>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/everest,es8389.yaml b/Documentation/devicetree/bindings/sound/everest,es8389.yaml new file mode 100644 index 000000000..7db84cf11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8389.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8389.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8389 audio CODEC + +maintainers: + - Michael Zhang <zhangyi@everest-semi.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: everest,es8389 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + "#sound-dai-cells": + const: 0 + + vdda-supply: + description: + Analogue power supply. + + vddd-supply: + description: + Interface power supply. + +required: + - compatible + - reg + - "#sound-dai-cells" + - vddd-supply + - vdda-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8389: codec@10 { + compatible = "everest,es8389"; + reg = <0x10>; + #sound-dai-cells = <0>; + vddd-supply = <&vdd3v3>; + vdda-supply = <&vdd3v3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/foursemi,fs2105s.yaml b/Documentation/devicetree/bindings/sound/foursemi,fs2105s.yaml new file mode 100644 index 000000000..4da735317 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/foursemi,fs2105s.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/foursemi,fs2105s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: FourSemi FS2104/5S Digital Audio Amplifier + +maintainers: + - Nick Li <nick.li@foursemi.com> + +description: + The FS2104 is a 15W Inductor-Less, Stereo, Closed-Loop, + Digital Input Class-D Power Amplifier with Enhanced Signal Processing. + The FS2105S is a 30W Inductor-Less, Stereo, Closed-Loop, + Digital Input Class-D Power Amplifier with Enhanced Signal Processing. + +properties: + compatible: + oneOf: + - items: + - enum: + - foursemi,fs2104 + - const: foursemi,fs2105s + - enum: + - foursemi,fs2105s + + reg: + maxItems: 1 + + clocks: + items: + - description: The clock of I2S BCLK + + clock-names: + items: + - const: bclk + + interrupts: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + pvdd-supply: + description: + Regulator for power supply(PVDD in datasheet). + + dvdd-supply: + description: + Regulator for digital supply(DVDD in datasheet). + + reset-gpios: + maxItems: 1 + description: + It's the SDZ pin in datasheet, the pin is active low, + it will power down and reset the chip to shut down state. + + firmware-name: + maxItems: 1 + description: | + The firmware(*.bin) contains: + a. Register initialization settings + b. DSP effect parameters + c. Multi-scene sound effect configurations(optional) + It's gernerated by FourSemi's tuning tool. + +required: + - compatible + - reg + - '#sound-dai-cells' + - pvdd-supply + - dvdd-supply + - reset-gpios + - firmware-name + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@68 { + compatible = "foursemi,fs2105s"; + reg = <0x68>; + clocks = <&clocks 18>; + clock-names = "bclk"; + #sound-dai-cells = <0>; + pvdd-supply = <&pvdd_supply>; + dvdd-supply = <&dvdd_supply>; + reset-gpios = <&gpio 18 GPIO_ACTIVE_LOW>; + firmware-name = "fs2105s-btl-2p0-0s.bin"; + pinctrl-names = "default"; + pinctrl-0 = <&fs210x_pins_default>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000..aa4be7170 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.yaml b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml new file mode 100644 index 000000000..07b9a3876 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml @@ -0,0 +1,154 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Mixer (AUDMIX). + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Frank Li <Frank.Li@nxp.com> + +description: | + 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. + +properties: + compatible: + enum: + - fsl,imx8qm-audmix + - fsl,imx952-audmix + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: ipg + + power-domains: + maxItems: 1 + + dais: + description: contain a list of phandles to AUDMIX connected DAIs. + $ref: /schemas/types.yaml#/definitions/phandle-array + minItems: 2 + items: + - description: the AUDMIX output + maxItems: 1 + - description: serial audio input 1 + maxItems: 1 + - description: serial audio input 2 + maxItems: 1 + + ports: + $ref: /schemas/graph.yaml#/properties/ports + patternProperties: + '^port@[0-1]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Input port from SAI TX + + properties: + port@2: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Output port to SAI RX + +required: + - compatible + - reg + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-audmix + then: + required: + - power-domains + +unevaluatedProperties: false + +examples: + - | + audmix@59840000 { + compatible = "fsl,imx8qm-audmix"; + reg = <0x59840000 0x10000>; + clocks = <&amix_lpcg 0>; + clock-names = "ipg"; + power-domains = <&pd_audmix>; + dais = <&sai4>, <&sai5>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0>; + playback-only; + + amix_endpoint0: endpoint { + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + dai-tdm-slot-width-map = <32 8 32>; + dai-format = "dsp_a"; + remote-endpoint = <&be00_ep>; + }; + }; + + port@1 { + reg = <1>; + playback-only; + + amix_endpoint1: endpoint { + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + dai-tdm-slot-width-map = <32 8 32>; + dai-format = "dsp_a"; + remote-endpoint = <&be01_ep>; + }; + }; + + port@2 { + reg = <2>; + capture-only; + + amix_endpoint2: endpoint { + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + dai-tdm-slot-width-map = <32 8 32>; + dai-format = "dsp_a"; + bitclock-master; + frame-master; + remote-endpoint = <&be02_ep>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml new file mode 100644 index 000000000..d5727f8bf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,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: + items: + - 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 + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + easrc: easrc@300c0000 { + compatible = "fsl,imx8mn-easrc"; + reg = <0x300c0000 0x10000>; + interrupts = <0x0 122 0x4>; + clocks = <&clk IMX8MN_CLK_ASRC_ROOT>; + clock-names = "mem"; + dmas = <&sdma2 16 23 0> , <&sdma2 17 23 0>, + <&sdma2 18 23 0> , <&sdma2 19 23 0>, + <&sdma2 20 23 0> , <&sdma2 21 23 0>, + <&sdma2 22 23 0> , <&sdma2 23 23 0>; + dma-names = "ctx0_rx", "ctx0_tx", + "ctx1_rx", "ctx1_tx", + "ctx2_rx", "ctx2_tx", + "ctx3_rx", "ctx3_tx"; + firmware-name = "imx/easrc/easrc-imx8mn.bin"; + fsl,asrc-rate = <8000>; + fsl,asrc-format = <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml new file mode 100644 index 000000000..27c34ce4c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -0,0 +1,136 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,esai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Enhanced Serial Audio Interface (ESAI) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Frank Li <Frank.Li@nxp.com> + +description: + 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. + +properties: + compatible: + oneOf: + - enum: + - fsl,imx35-esai + - fsl,imx6ull-esai + - fsl,vf610-esai + - items: + - enum: + - fsl,imx8qm-esai + - const: fsl,imx6ull-esai + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 3 + items: + - description: + The core clock used to access registers. + - description: + The esai baud clock for esai controller used to + derive HCK, SCK and FS. + - description: + The system clock derived from ahb clock used to + derive HCK, SCK and FS. + - description: + 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. + + clock-names: + minItems: 3 + items: + - const: core + - const: extal + - const: fsys + - const: spba + + dmas: + minItems: 2 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + power-domains: + maxItems: 1 + + fsl,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 64 + description: + 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: + $ref: /schemas/types.yaml#/definitions/flag + description: + 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. + + 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. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: fsl,imx8qm-esai + then: + required: + - power-domains + else: + properties: + power-domains: false + +examples: + - | + esai@2024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + big-endian; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml new file mode 100644 index 000000000..608defc93 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml @@ -0,0 +1,190 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx-asrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Asynchronous Sample Rate Converter (ASRC) Controller + +description: + The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of + a signal associated with an input clock into a signal associated with a + different output clock. The driver currently works as a Front End of DPCM + with other Back Ends Audio controller such as ESAI, SSI and SAI. It has + three pairs to support three substreams within totally 10 channels. + +maintainers: + - Shawn Guo <shawnguo@kernel.org> + - Sascha Hauer <s.hauer@pengutronix.de> + +properties: + compatible: + oneOf: + - enum: + - fsl,imx35-asrc + - fsl,imx53-asrc + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + - fsl,imx952-asrc + - items: + - enum: + - fsl,imx6sx-asrc + - fsl,imx6ul-asrc + - const: fsl,imx53-asrc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 6 + + dma-names: + items: + - const: rxa + - const: rxb + - const: rxc + - const: txa + - const: txb + - const: txc + + clocks: + maxItems: 19 + + clock-names: + items: + - const: mem + - const: ipg + - const: asrck_0 + - const: asrck_1 + - const: asrck_2 + - const: asrck_3 + - const: asrck_4 + - const: asrck_5 + - const: asrck_6 + - const: asrck_7 + - const: asrck_8 + - const: asrck_9 + - const: asrck_a + - const: asrck_b + - const: asrck_c + - const: asrck_d + - const: asrck_e + - const: asrck_f + - const: spba + + power-domains: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + fsl,asrc-rate: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The mutual sample rate used by DPCM Back Ends + + fsl,asrc-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The mutual sample width used by DPCM Back Ends + enum: [16, 24] + + fsl,asrc-clk-map: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Defines clock map used in driver + <0> - select the map for asrc0 in imx8qm/imx8qxp + <1> - select the map for asrc1 in imx8qm/imx8qxp + enum: [0, 1] + + big-endian: + type: boolean + description: + If this property is absent, the little endian mode will be in use as + default. Otherwise, the big endian mode will be in use for all the + device registers. + + fsl,asrc-format: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Defines a mutual sample format used by DPCM Back Ends, which can + replace the fsl,asrc-width. The value is 2 (S16_LE), or 6 (S24_LE). + enum: [2, 6] + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + - fsl,asrc-rate + - fsl,asrc-width + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + then: + required: + - fsl,asrc-clk-map + else: + properties: + fsl,asrc-clk-map: false + + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + then: + required: + - power-domains + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/clock/imx6qdl-clock.h> + asrc: asrc@2034000 { + compatible = "fsl,imx53-asrc"; + reg = <0x02034000 0x4000>; + interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX6QDL_CLK_ASRC_IPG>, + <&clks IMX6QDL_CLK_ASRC_MEM>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks IMX6QDL_CLK_ASRC>, <&clks 0>, <&clks 0>, + <&clks IMX6QDL_CLK_SPBA>; + clock-names = "mem", "ipg", "asrck_0", + "asrck_1", "asrck_2", "asrck_3", "asrck_4", + "asrck_5", "asrck_6", "asrck_7", "asrck_8", + "asrck_9", "asrck_a", "asrck_b", "asrck_c", + "asrck_d", "asrck_e", "asrck_f", "spba"; + dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, + <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; + dma-names = "rxa", "rxb", "rxc", + "txa", "txb", "txc"; + fsl,asrc-rate = <48000>; + fsl,asrc-width = <16>; + + port { + playback-only; + + asrc_endpoint: endpoint { + remote-endpoint = <&fe00_ep>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml new file mode 100644 index 000000000..5eb6f5812 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml @@ -0,0 +1,111 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx-audio-es8328.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale i.MX audio complex with ES8328 codec + +maintainers: + - Shawn Guo <shawnguo@kernel.org> + - Sascha Hauer <s.hauer@pengutronix.de> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: fsl,imx-audio-es8328 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex + + ssi-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the i.MX SSI controller + + jack-gpio: + description: Optional GPIO for headphone jack + maxItems: 1 + + audio-amp-supply: + description: Power regulator for speaker amps + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to the ES8328 audio codec + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the second + being the connection's source. 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: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The internal port of the i.MX audio muxer (AUDMUX) + enum: [1, 2, 7] + default: 1 + + mux-ext-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The external port of the i.MX audio muxer (AUDMIX) + enum: [3, 4, 5, 6] + default: 3 + +required: + - compatible + - model + - ssi-controller + - jack-gpio + - audio-amp-supply + - audio-codec + - audio-routing + - mux-int-port + - mux-ext-port + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "fsl,imx-audio-es8328"; + model = "imx-audio-es8328"; + ssi-controller = <&ssi1>; + audio-codec = <&codec>; + jack-gpio = <&gpio5 15 0>; + audio-amp-supply = <®_audio_amp>; + audio-routing = + "Speaker", "LOUT2", + "Speaker", "ROUT2", + "Speaker", "audio-amp", + "Headphone", "ROUT1", + "Headphone", "LOUT1", + "LINPUT1", "Mic Jack", + "RINPUT1", "Mic Jack", + "Mic Jack", "Mic Bias"; + mux-int-port = <1>; + mux-ext-port = <3>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,imx95-cm7-sof.yaml b/Documentation/devicetree/bindings/sound/fsl,imx95-cm7-sof.yaml new file mode 100644 index 000000000..f00ae3219 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx95-cm7-sof.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx95-cm7-sof.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP imx95 CM7 core + +maintainers: + - Daniel Baluta <daniel.baluta@nxp.com> + +description: NXP imx95 CM7 core used for audio processing + +properties: + compatible: + const: fsl,imx95-cm7-sof + + reg: + maxItems: 1 + + reg-names: + const: sram + + memory-region: + maxItems: 1 + + memory-region-names: + const: dma + + port: + description: SAI3 port + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - reg-names + - memory-region + - memory-region-names + - port + +allOf: + - $ref: fsl,sof-cpu.yaml# + +unevaluatedProperties: false + +examples: + - | + cm7-cpu@80000000 { + compatible = "fsl,imx95-cm7-sof"; + reg = <0x80000000 0x6100000>; + reg-names = "sram"; + mboxes = <&mu7 2 0>, <&mu7 2 1>, <&mu7 3 0>, <&mu7 3 1>; + mbox-names = "txdb0", "txdb1", "rxdb0", "rxdb1"; + memory-region = <&adma_res>; + memory-region-names = "dma"; + port { + /* SAI3-WM8962 link */ + endpoint { + remote-endpoint = <&wm8962_ep>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml new file mode 100644 index 000000000..4c7dadb31 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,micfil.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP MICFIL Digital Audio Interface (MICFIL) + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + The MICFIL digital interface provides a 16-bit or 24-bit audio signal + from a PDM microphone bitstream in a configurable output sampling rate. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx95-micfil + - const: fsl,imx93-micfil + + - enum: + - fsl,imx8mm-micfil + - fsl,imx8mp-micfil + - fsl,imx93-micfil + - fsl,imx943-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 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + micfil: audio-controller@30080000 { + compatible = "fsl,imx8mm-micfil"; + reg = <0x30080000 0x10000>; + interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 44 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 45 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>; + clock-names = "ipg_clk", "ipg_clk_app"; + dmas = <&sdma2 24 25 0>; + dma-names = "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.yaml b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml new file mode 100644 index 000000000..bcc265a74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml @@ -0,0 +1,125 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,mqs.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Medium Quality Sound (MQS) + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Chancel Liu <chancel.liu@nxp.com> + +description: | + Medium quality sound (MQS) is used to generate medium quality audio + via a standard GPIO in the pinmux, allowing the user to connect + stereo speakers or headphones to a power amplifier without an + additional DAC chip. + +properties: + compatible: + enum: + - fsl,imx6sx-mqs + - fsl,imx8qm-mqs + - fsl,imx8qxp-mqs + - fsl,imx93-mqs + - fsl,imx943-aonmix-mqs + - fsl,imx943-wakeupmix-mqs + - fsl,imx95-aonmix-mqs + - fsl,imx95-netcmix-mqs + + "#sound-dai-cells": + const: 0 + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + minItems: 1 + maxItems: 2 + + gpr: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to the General Purpose Register (GPR) node + + reg: + maxItems: 1 + + power-domains: + maxItems: 1 + + resets: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - fsl,imx6sx-mqs + - fsl,imx93-mqs + then: + required: + - gpr + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-mqs + - fsl,imx8qxp-mqs + then: + properties: + clocks: + items: + - description: Master clock + - description: Clock for register access + clock-names: + items: + - const: mclk + - const: core + required: + - reg + - power-domains + else: + properties: + clocks: + items: + - description: Master clock + clock-names: + items: + - const: mclk + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/imx6sx-clock.h> + mqs0: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + }; + + - | + #include <dt-bindings/firmware/imx/rsrc.h> + mqs1: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&mqs0_lpcg 0>, <&mqs0_lpcg 1>; + clock-names = "mclk", "core"; + power-domains = <&pd IMX_SC_R_MQS_0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml b/Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml new file mode 100644 index 000000000..d12774b42 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,mxs-audio-sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale MXS audio complex with SGTL5000 codec + +maintainers: + - Frank Li <Frank.Li@nxp.com> + +properties: + compatible: + items: + - enum: + - bluegiga,apx4devkit-sgtl5000 + - denx,m28evk-sgtl5000 + - fsl,imx28-evk-sgtl5000 + - fsl,imx28-mbmx28lc-sgtl5000 + - fsl,imx28-tx28-sgtl5000 + - const: fsl,mxs-audio-sgtl5000 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex + + saif-controllers: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: The phandle list of the MXS SAIF controller + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SGTL5000 audio codec + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. 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 + +required: + - compatible + - saif-controllers + - audio-codec + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "fsl,imx28-evk-sgtl5000", "fsl,mxs-audio-sgtl5000"; + model = "imx28-evk-sgtl5000"; + saif-controllers = <&saif0 &saif1>; + audio-codec = <&sgtl5000>; + audio-routing = + "MIC_IN", "Mic Jack", + "Mic Jack", "Mic Bias", + "Headphone Jack", "HP_OUT"; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml b/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml new file mode 100644 index 000000000..a23e49198 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml @@ -0,0 +1,147 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 several DAIs. For each DAI, the DAI is working in interleaved mode + if only one QMC channel is used by the DAI or it is working in non-interleaved + mode if several QMC channels are used by the DAI. + +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 + minItems: 1 + description: + Should be a phandle/number pair list. The list of phandle to QMC node + and the QMC channel pair to use for this DAI. + If only one phandle/number pair is provided, this DAI works in + interleaved mode, i.e. audio channels for this DAI are interleaved in + the QMC channel. If more than one pair is provided, this DAI works + in non-interleave mode. In that case the first audio channel uses the + the first QMC channel, the second audio channel uses the second QMC + channel, etc... + + 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>; + }; + dai@18 { + reg = <18>; + /* Non-interleaved mode */ + fsl,qmc-chan = <&qmc 18>, <&qmc 19>; + }; + }; + + 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>; + }; + }; + simple-audio-card,dai-link@2 { + reg = <2>; + format = "dsp_b"; + cpu { + sound-dai = <&audio_controller 18>; + }; + codec { + sound-dai = <&codec3>; + dai-tdm-slot-num = <2>; + dai-tdm-slot-width = <8>; + /* TS 9, 10 */ + dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0 0 1 1>; + dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0 0 1 1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml new file mode 100644 index 000000000..e3cadb93c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -0,0 +1,144 @@ +# 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: + oneOf: + - 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 + - fsl,imx95-rpmsg-audio + - items: + - enum: + - fsl,imx94-rpmsg-audio + - fsl,imx952-rpmsg-audio + - const: fsl,imx95-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: + $ref: /schemas/types.yaml#/definitions/string + description: | + A string property to assign rpmsg channel this sound card sits on. + This property can be omitted if there is only one sound card and it sits + on "rpmsg-audio-channel". + enum: + - rpmsg-audio-channel + - rpmsg-micfil-channel + + hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/imx8mn-clock.h> + + rpmsg_audio: rpmsg_audio { + compatible = "fsl,imx8mn-rpmsg-audio"; + model = "wm8524-audio"; + fsl,enable-lpa; + fsl,rpmsg-out; + clocks = <&clk IMX8MN_CLK_SAI3_IPG>, + <&clk IMX8MN_CLK_SAI3_ROOT>, + <&clk IMX8MN_CLK_SDMA3_ROOT>, + <&clk IMX8MN_AUDIO_PLL1_OUT>, + <&clk IMX8MN_AUDIO_PLL2_OUT>; + clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; + }; + + - | + #include <dt-bindings/clock/imx8mm-clock.h> + + rpmsg_micfil: audio-controller { + compatible = "fsl,imx8mm-rpmsg-audio"; + model = "micfil-audio"; + fsl,rpmsg-channel-name = "rpmsg-micfil-channel"; + fsl,enable-lpa; + fsl,rpmsg-in; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>, + <&clk IMX8MM_CLK_SDMA3_ROOT>, + <&clk IMX8MM_AUDIO_PLL1_OUT>, + <&clk IMX8MM_AUDIO_PLL2_OUT>; + clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000..ba65b3f3d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,286 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx6ul-sai + - fsl,imx7d-sai + - const: fsl,imx6sx-sai + + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + - items: + - enum: + - fsl,imx6sx-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - fsl,imx93-sai + - fsl,imx95-sai + - fsl,vf610-sai + - items: + - enum: + - fsl,imx94-sai + - fsl,imx952-sai + - const: fsl,imx95-sai + + reg: + maxItems: 1 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 5 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + power-domains: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + items: + - enum: [ rx, tx ] + - const: tx + + interrupts: + items: + - description: receive and transmit interrupt + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: port for TX and RX + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: port for TX only + + port@2: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: port for RX only + + 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-amix-mode: + $ref: /schemas/types.yaml#/definitions/string + description: + The audmix module is bypassed from hardware or not. + enum: [none, bypass, audmix] + default: none + + 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,sai-bit-clock-swap: + description: + Enable Bit Clock Swap, which swaps the bit clock used by the transmitter + or receiver in asynchronous mode, i.e. makes transmitter use RX_BCLK and + TX_SYNC, and vice versa, makes receiver use TX_BCLK and RX_SYNC. + 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 + + - if: + required: + - fsl,sai-amix-mode + then: + properties: + compatible: + contains: + const: fsl,imx952-sai + +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>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@1 { + reg = <1>; + playback-only; + + sai1_endpoint0: endpoint { + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + dai-tdm-slot-width-map = <32 8 32>; + dai-format = "dsp_a"; + bitclock-master; + frame-master; + remote-endpoint = <&mcodec01_ep>; + }; + }; + + port@2 { + reg = <2>; + capture-only; + + sai1_endpoint1: endpoint { + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + dai-tdm-slot-width-map = <32 8 32>; + dai-format = "dsp_a"; + remote-endpoint = <&fe02_ep>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,saif.yaml b/Documentation/devicetree/bindings/sound/fsl,saif.yaml new file mode 100644 index 000000000..0b5db6bb1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,saif.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/fsl,saif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale MXS Serial Audio Interface (SAIF) + +maintainers: + - Lukasz Majewski <lukma@denx.de> + +allOf: + - $ref: dai-common.yaml# + +description: + The SAIF is based on I2S module that is used to communicate with audio codecs, + but only with half-duplex manner (i.e. it can either transmit or receive PCM + audio). + +properties: + compatible: + const: fsl,imx28-saif + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + const: rx-tx + + "#clock-cells": + description: Configure the I2S device as MCLK clock provider. + const: 0 + + clocks: + maxItems: 1 + + fsl,saif-master: + description: Indicate that saif is a slave and its phandle points to master + $ref: /schemas/types.yaml#/definitions/phandle + +required: + - compatible + - reg + - "#sound-dai-cells" + - interrupts + - dmas + - dma-names + - clocks + +unevaluatedProperties: false + +examples: + - | + saif0: saif@80042000 { + compatible = "fsl,imx28-saif"; + reg = <0x80042000 2000>; + #sound-dai-cells = <0>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; + #clock-cells = <0>; + clocks = <&clks 53>; + }; + - | + saif1: saif@80046000 { + compatible = "fsl,imx28-saif"; + reg = <0x80046000 2000>; + #sound-dai-cells = <0>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; + clocks = <&clks 53>; + fsl,saif-master = <&saif0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,sgtl5000.yaml b/Documentation/devicetree/bindings/sound/fsl,sgtl5000.yaml new file mode 100644 index 000000000..c6ab1ca16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sgtl5000.yaml @@ -0,0 +1,113 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale SGTL5000 Stereo Codec + +maintainers: + - Fabio Estevam <festevam@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: fsl,sgtl5000 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + + clocks: + items: + - description: the clock provider of SYS_MCLK + + VDDA-supply: + description: the regulator provider of VDDA + + VDDIO-supply: + description: the regulator provider of VDDIO + + VDDD-supply: + description: the regulator provider of VDDD + + micbias-resistor-k-ohms: + description: The bias resistor to be used in kOhms. The resistor can take + values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not + mentioned or if the value is unknown, then micbias resistor is set to + 4k. + enum: [ 0, 2, 4, 8 ] + + micbias-voltage-m-volts: + description: The bias voltage to be used in mVolts. The voltage can take + values from 1.25V to 3V by 250mV steps. If this node is not mentioned + or the value is unknown, then the value is set to 1.25V. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] + + lrclk-strength: + description: | + The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1, 2, 3 ] + + sclk-strength: + description: | + The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1, 2, 3 ] + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <®_3p3v>; + VDDIO-supply = <®_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/fsl,sof-cpu.yaml b/Documentation/devicetree/bindings/sound/fsl,sof-cpu.yaml new file mode 100644 index 000000000..31863932d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sof-cpu.yaml @@ -0,0 +1,27 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sof-cpu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP audio processor common properties + +maintainers: + - Daniel Baluta <daniel.baluta@nxp.com> + +properties: + mboxes: + maxItems: 4 + + mbox-names: + items: + - const: txdb0 + - const: txdb1 + - const: rxdb0 + - const: rxdb1 + +required: + - mboxes + - mbox-names + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 000000000..5654e9f61 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -0,0 +1,160 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + oneOf: + - items: + - enum: + - fsl,imx35-spdif + - fsl,imx6sx-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif + - fsl,imx8mq-spdif + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8ulp-spdif + - fsl,vf610-spdif + - items: + - enum: + - fsl,imx6sl-spdif + - fsl,imx6sx-spdif + - const: fsl,imx35-spdif + + reg: + maxItems: 1 + + interrupts: + minItems: 1 + items: + - description: Combined or receive interrupt + - description: Transmit interrupt + + 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: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +allOf: + - if: + properties: + compatible: + enum: + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + then: + properties: + interrupts: + minItems: 2 + else: + properties: + interrupts: + maxItems: 1 + + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + then: + required: + - power-domains + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.yaml b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml new file mode 100644 index 000000000..4ab10cd3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml @@ -0,0 +1,194 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Serial Interface + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: + 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. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx50-ssi + - fsl,imx53-ssi + - const: fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx25-ssi + - fsl,imx27-ssi + - fsl,imx35-ssi + - fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx6q-ssi + - fsl,imx6sl-ssi + - fsl,imx6sx-ssi + - const: fsl,imx51-ssi + - items: + - const: fsl,imx21-ssi + - items: + - const: fsl,mpc8610-ssi + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: The ipg clock for register access + - description: clock for SSI master mode + minItems: 1 + + clock-names: + items: + - const: ipg + - const: baud + minItems: 1 + + dmas: + oneOf: + - items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + - items: + - description: DMA controller phandle and request line for RX0 + - description: DMA controller phandle and request line for TX0 + - description: DMA controller phandle and request line for RX1 + - description: DMA controller phandle and request line for TX1 + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - items: + - const: rx0 + - const: tx0 + - const: rx1 + - const: tx1 + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + + cell-index: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: The SSI index + + ac97-gpios: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: Please refer to soc-ac97link.txt + + codec-handle: + $ref: /schemas/types.yaml#/definitions/phandle + description: + 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,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for SFCSR[TFWM0]. + enum: [8, 15] + + fsl,fiq-stream-filter: + type: boolean + description: + 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. + + fsl,mode: + $ref: /schemas/types.yaml#/definitions/string + enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master, + lj-slave, lj-master, rj-slave, rj-master ] + description: | + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - Left justified mode, SSI is clock slave + "lj-master" - Left justified mode, SSI is clock master + "rj-slave" - Right justified mode, SSI is clock slave + "rj-master" - Right justified mode, SSI is clock master + + fsl,ssi-asynchronous: + type: boolean + description: 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: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + playback of audio. This is typically dictated by SOC + design. Only used on Power Architecture. + + fsl,capture-dma: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + capture (recording) of audio. This is typically dictated + by SOC design. Only used on Power Architecture. + +required: + - compatible + - reg + - interrupts + - fsl,fifo-depth + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx6qdl-clock.h> + ssi@2028000 { + compatible = "fsl,imx6q-ssi", "fsl,imx51-ssi"; + reg = <0x02028000 0x4000>; + interrupts = <GIC_SPI 46 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX6QDL_CLK_SSI1_IPG>, + <&clks IMX6QDL_CLK_SSI1>; + clock-names = "ipg", "baud"; + dmas = <&sdma 37 1 0>, <&sdma 38 1 0>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + fsl,fifo-depth = <15>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml new file mode 100644 index 000000000..f68d0e0ec --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml @@ -0,0 +1,160 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + - fsl,imx95-xcvr + + reg: + items: + - description: 20K RAM for code and data + - description: registers space + - description: RX FIFO address + - description: TX FIFO address + + reg-names: + items: + - const: ram + - const: regs + - const: rxfifo + - const: txfifo + + interrupts: + items: + - description: WAKEUPMIX Audio XCVR Interrupt 1 + - description: WAKEUPMIX Audio XCVR Interrupt 2 + - description: SPDIF wakeup interrupt from PHY + minItems: 1 + + clocks: + items: + - description: Peripheral clock + - description: PHY clock + - description: SPBA clock + - description: PLL clock + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + items: + - const: ipg + - const: phy + - const: spba + - const: pll_ipg + - 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 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - reg-names + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: fsl,imx8mp-xcvr + then: + required: + - resets + + - if: + properties: + compatible: + contains: + enum: + - fsl,imx93-xcvr + - fsl,imx95-xcvr + then: + properties: + interrupts: + minItems: 2 + maxItems: 2 + else: + properties: + interrupts: + minItems: 3 + maxItems: 3 + + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8mp-xcvr + - fsl,imx93-xcvr + then: + properties: + clocks: + maxItems: 4 + clock-names: + maxItems: 4 + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + #include <dt-bindings/reset/imx8mp-reset.h> + + xcvr: xcvr@30cc0000 { + compatible = "fsl,imx8mp-xcvr"; + reg = <0x30cc0000 0x800>, + <0x30cc0800 0x400>, + <0x30cc0c00 0x080>, + <0x30cc0e00 0x080>; + reg-names = "ram", "regs", "rxfifo", "txfifo"; + interrupts = <GIC_SPI 128 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 129 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 146 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_IPG>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_PHY>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_SPBA2_ROOT>, + <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUDPLL_ROOT>; + clock-names = "ipg", "phy", "spba", "pll_ipg"; + dmas = <&sdma2 30 2 0>, <&sdma2 31 2 0>; + dma-names = "rx", "tx"; + resets = <&audiomix_reset 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml new file mode 100644 index 000000000..88eb20bb0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -0,0 +1,243 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl-asoc-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Generic ASoC Sound Card with ASRC support + +description: + 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 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. + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx-sgtl5000 + - fsl,imx25-pdk-sgtl5000 + - fsl,imx53-cpuvo-sgtl5000 + - fsl,imx51-babbage-sgtl5000 + - fsl,imx53-m53evk-sgtl5000 + - fsl,imx53-qsb-sgtl5000 + - fsl,imx53-voipac-sgtl5000 + - fsl,imx6-armadeus-sgtl5000 + - fsl,imx6-rex-sgtl5000 + - fsl,imx6-sabreauto-cs42888 + - fsl,imx6-wandboard-sgtl5000 + - fsl,imx6dl-nit6xlite-sgtl5000 + - fsl,imx6q-ba16-sgtl5000 + - fsl,imx6q-nitrogen6_max-sgtl5000 + - fsl,imx6q-nitrogen6_som2-sgtl5000 + - fsl,imx6q-nitrogen6x-sgtl5000 + - fsl,imx6q-sabrelite-sgtl5000 + - fsl,imx6q-sabresd-wm8962 + - fsl,imx6q-udoo-ac97 + - fsl,imx6q-ventana-sgtl5000 + - fsl,imx6sl-evk-wm8962 + - fsl,imx6sx-sdb-mqs + - fsl,imx6sx-sdb-wm8962 + - fsl,imx7d-evk-wm8960 + - karo,tx53-audio-sgtl5000 + - tq,imx53-mba53-sgtl5000 + - enum: + - fsl,imx-audio-ac97 + - fsl,imx-audio-cs42888 + - fsl,imx-audio-mqs + - fsl,imx-audio-sgtl5000 + - fsl,imx-audio-wm8960 + - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif + - items: + - enum: + - fsl,imx-audio-ac97 + - fsl,imx-audio-cs42888 + - fsl,imx-audio-cs427x + - fsl,imx-audio-mqs + - fsl,imx-audio-nau8822 + - fsl,imx-audio-sgtl5000 + - fsl,imx-audio-si476x + - fsl,imx-audio-tlv320 + - fsl,imx-audio-tlv320aic31xx + - fsl,imx-audio-tlv320aic32x4 + - fsl,imx-audio-wm8524 + - fsl,imx-audio-wm8904 + - fsl,imx-audio-wm8960 + - fsl,imx-audio-wm8962 + - fsl,imx-audio-wm8958 + - fsl,imx-audio-spdif + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex + + audio-asrc: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter, + spdif_receiver or both. + minItems: 1 + maxItems: 2 + items: + maxItems: 1 + + audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + + 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. There're a few pre-designed board + connectors. "AMIC" stands for Analog Microphone Jack. + "DMIC" stands for Digital Microphone Jack. The "Mic Jack" and "AMIC" + are redundant while coexisting in order to support the old bindings + of wm8962 and sgtl5000. + + hp-det-gpio: + deprecated: true + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + mic-det-gpio: + deprecated: true + maxItems: 1 + description: The GPIO that detect microphones are plugged in + + mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphones are plugged in + + bitclock-master: + $ref: simple-card.yaml#/definitions/bitclock-master + description: Indicates dai-link bit clock master. + + frame-master: + $ref: simple-card.yaml#/definitions/frame-master + description: Indicates dai-link frame master. + + format: + $ref: simple-card.yaml#/definitions/format + description: audio format. + + frame-inversion: + $ref: simple-card.yaml#/definitions/frame-inversion + description: dai-link uses frame clock inversion. + + bitclock-inversion: + $ref: simple-card.yaml#/definitions/bitclock-inversion + description: dai-link uses bit clock inversion. + + mclk-id: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. + minItems: 1 + maxItems: 2 + + mux-int-port: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [1, 2, 7] + description: The internal port of the i.MX audio muxer (AUDMUX) + + mux-ext-port: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 4, 5, 6] + description: The external port of the i.MX audio muxer + + ssi-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + description: The phandle of an S/PDIF CPU DAI controller. + + spdif-out: + type: boolean + deprecated: true + description: | + If present, the transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector or jack on the + board or it's connecting to some other IP block, such as an HDMI + encoder or display-controller. + + spdif-in: + type: boolean + deprecated: true + description: | + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + +required: + - compatible + - model + +unevaluatedProperties: false + +examples: + - | + sound-cs42888 { + compatible = "fsl,imx-audio-cs42888"; + model = "cs42888-audio"; + audio-cpu = <&esai>; + audio-asrc = <&asrc>; + audio-codec = <&cs42888>; + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "Line In Jack", + "AIN1R", "Line In Jack", + "AIN2L", "Line In Jack", + "AIN2R", "Line In Jack"; + }; + + - | + sound-spdif-asrc { + compatible = "fsl,imx-audio-spdif"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + }; diff --git a/Documentation/devicetree/bindings/sound/google,chv3-codec.yaml b/Documentation/devicetree/bindings/sound/google,chv3-codec.yaml new file mode 100644 index 000000000..5329dc140 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,chv3-codec.yaml @@ -0,0 +1,31 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,chv3-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google Chameleon v3 audio codec + +maintainers: + - PaweÅ‚ Anikiel <pan@semihalf.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: google,chv3-codec + + "#sound-dai-cells": + const: 0 + +required: + - compatible + +additionalProperties: false + +examples: + - | + audio-codec { + compatible = "google,chv3-codec"; + }; diff --git a/Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml b/Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml new file mode 100644 index 000000000..3ce910f44 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,chv3-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google Chameleon v3 I2S device + +maintainers: + - PaweÅ‚ Anikiel <pan@semihalf.com> + +description: | + I2S device for the Google Chameleon v3. The device handles both RX + and TX using a producer/consumer ring buffer design. + +properties: + compatible: + const: google,chv3-i2s + + reg: + items: + - description: core registers + - description: irq registers + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@c0060300 { + compatible = "google,chv3-i2s"; + reg = <0xc0060300 0x100>, + <0xc0060f00 0x10>; + interrupts = <GIC_SPI 20 IRQ_TYPE_LEVEL_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml new file mode 100644 index 000000000..dd51e8c5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -0,0 +1,78 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$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/embedded-controller/google,cros-ec.yaml). + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: google,cros-ec-codec + + "#sound-dai-cells": + const: 1 + + reg: + items: + - description: | + Physical base address and length of shared memory region from EC. + It contains 3 unsigned 32-bit integer. The first 2 integers + combine to become an unsigned 64-bit physical address. + The last one integer is the length of the shared memory. + + memory-region: + maxItems: 1 + description: | + Shared memory region to EC. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + +required: + - compatible + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + reserved_mem: reserved-mem@52800000 { + compatible = "shared-dma-pool"; + reg = <0x52800000 0x100000>; + no-map; + }; + spi { + #address-cells = <1>; + #size-cells = <0>; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + reg = <0>; + interrupts = <93 0>; + + codecs { + #address-cells = <2>; + #size-cells = <1>; + + cros_ec_codec: ec-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,goldfish-audio.yaml b/Documentation/devicetree/bindings/sound/google,goldfish-audio.yaml new file mode 100644 index 000000000..d395a5cbc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,goldfish-audio.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,goldfish-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Android Goldfish Audio + +maintainers: + - Kuan-Wei Chiu <visitorckw@gmail.com> + +description: + Android goldfish audio device generated by Android emulator. + +properties: + compatible: + const: google,goldfish-audio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + sound@9030000 { + compatible = "google,goldfish-audio"; + reg = <0x9030000 0x100>; + interrupts = <4>; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml new file mode 100644 index 000000000..bac940553 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7180-trogdor.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7180-Trogdor ASoC sound card driver + +maintainers: + - Rohit kumar <quic_rohkumar@quicinc.com> + - Cheng-Yi Chiang <cychiang@chromium.org> + +description: + This binding describes the SC7180 sound card which uses LPASS for audio. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - google,sc7180-trogdor + - google,sc7180-coachz + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + dmic-gpios: + maxItems: 1 + description: GPIO for switching between DMICs + +patternProperties: + "^dai-link(@[0-9])?$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 4 + + required: + - link-name + - cpu + - codec + + additionalProperties: false + +required: + - compatible + - "#address-cells" + - "#size-cells" + +unevaluatedProperties: false + +examples: + + - | + sound { + compatible = "google,sc7180-trogdor"; + model = "sc7180-rt5682-max98357a-2mic"; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR"; + + #address-cells = <1>; + #size-cells = <0>; + + dmic-gpios = <&tlmm 86 0>; + + dai-link@0 { + link-name = "MultiMedia0"; + reg = <0>; + cpu { + sound-dai = <&lpass_cpu 0>; + }; + + codec { + sound-dai = <&alc5682 0>; + }; + }; + + dai-link@1 { + link-name = "MultiMedia1"; + reg = <1>; + cpu { + sound-dai = <&lpass_cpu 1>; + }; + + codec { + sound-dai = <&max98357a>; + }; + }; + + dai-link@2 { + link-name = "MultiMedia2"; + reg = <2>; + cpu { + sound-dai = <&lpass_hdmi 0>; + }; + + codec { + sound-dai = <&msm_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml new file mode 100644 index 000000000..cdcd7c6f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,sc7280-herobrine.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google SC7280-Herobrine ASoC sound card driver + +maintainers: + - Judy Hsiao <judyhsiao@chromium.org> + +description: + This binding describes the SC7280 sound card which uses LPASS for audio. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - google,sc7280-herobrine + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + reg: + maxItems: 1 + description: dai link address. + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: true + + required: + - sound-dai + + additionalProperties: false + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: true + + required: + - sound-dai + + additionalProperties: false + + platform: + description: Holds subnode which includes the phandle of q6apm platform device. + type: object + properties: + sound-dai: + maxItems: 1 + + required: + - sound-dai + + additionalProperties: false + + required: + - link-name + - cpu + - codec + - reg + + additionalProperties: false + +required: + - compatible + - "#address-cells" + - "#size-cells" + +unevaluatedProperties: false + +examples: + + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound { + compatible = "google,sc7280-herobrine"; + model = "sc7280-wcd938x-max98360a-4dmic"; + + audio-routing = + "IN1_HPHL", "HPHL_OUT", + "IN2_HPHR", "HPHR_OUT", + "AMIC1", "MIC BIAS1", + "AMIC2", "MIC BIAS2", + "VA DMIC0", "MIC BIAS3", + "VA DMIC1", "MIC BIAS3", + "VA DMIC2", "MIC BIAS4", + "VA DMIC3", "MIC BIAS4", + "TX SWR_ADC0", "ADC1_OUTPUT", + "TX SWR_ADC1", "ADC2_OUTPUT", + "TX SWR_ADC2", "ADC3_OUTPUT", + "TX SWR_DMIC0", "DMIC1_OUTPUT", + "TX SWR_DMIC1", "DMIC2_OUTPUT", + "TX SWR_DMIC2", "DMIC3_OUTPUT", + "TX SWR_DMIC3", "DMIC4_OUTPUT"; + + #address-cells = <1>; + #size-cells = <0>; + + dai-link@0 { + link-name = "WCD Playback"; + reg = <LPASS_CDC_DMA_RX0>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_RX0>; + }; + + codec { + sound-dai = <&wcd938x 0>, <&swr0 0>, <&rxmacro 0>; + }; + }; + dai-link@1 { + link-name = "WCD Capture"; + reg = <LPASS_CDC_DMA_TX3>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_TX3>; + }; + + codec { + sound-dai = <&wcd938x 1>, <&swr1 0>, <&txmacro 0>; + }; + }; + + dai-link@2 { + link-name = "MI2S Playback"; + reg = <MI2S_SECONDARY>; + cpu { + sound-dai = <&lpass_cpu MI2S_SECONDARY>; + }; + + codec { + sound-dai = <&max98360a>; + }; + }; + + dai-link@3 { + link-name = "DMIC Capture"; + reg = <LPASS_CDC_DMA_VA_TX0>; + cpu { + sound-dai = <&lpass_cpu LPASS_CDC_DMA_VA_TX0>; + }; + + codec { + sound-dai = <&vamacro 0>; + }; + }; + + dai-link@5 { + link-name = "DP Playback"; + reg = <LPASS_DP_RX>; + cpu { + sound-dai = <&lpass_cpu LPASS_DP_RX>; + }; + + codec { + sound-dai = <&mdss_dp>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/gpio-audio-amp.yaml b/Documentation/devicetree/bindings/sound/gpio-audio-amp.yaml new file mode 100644 index 000000000..3690f3d16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gpio-audio-amp.yaml @@ -0,0 +1,270 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/gpio-audio-amp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio amplifier driven by GPIOs + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + Audio GPIO amplifiers are driven by GPIO in order to control the gain value + of the amplifier, its mute function and/or its bypass function. + + Those amplifiers are based on discrete components (analog switches, op-amps + and more) where some of them, mostly analog switches, are controlled by GPIOs + to adjust the gain value of the whole amplifier and/or to control + the mute and/or bypass function. + + For instance, the following piece of hardware is a GPIO amplifier + + +5VA + ^ + |\ | + | \ + Vin >---------------------------|+ \ + | +-------+-----> Vout + .--\/\/\/--+------------|- / | + | | | / | + v | |/ | | + GND o v | + \ GND | + gpio >-----------> \ | + o o | + | | | + | '--\/\/\/--. | + | +--\/\/\/--' + '---------------' + +properties: + compatible: + oneOf: + - const: gpio-audio-amp-mono + description: + A single channel amplifier. All features apply to this sole channel. + + - const: gpio-audio-amp-stereo + description: + A dual channel amplifier (left and right). All features apply to both + channels producing the same effect on both channels at the same time. + + vdd-supply: + description: Main power supply of the amplifier + + vddio-supply: + description: Power supply related to the control path + + vdda1-supply: + description: Analog power supply + + vdda2-supply: + description: Additional analog power supply + + mute-gpios: + description: GPIO to control the mute function + maxItems: 1 + + bypass-gpios: + description: GPIO to control the bypass function + maxItems: 1 + + gain-gpios: + description: | + GPIOs to control the amplifier gain + + The gain value is computed from GPIOs value from 0 to 2^N-1 with N the + number of GPIO described. The first GPIO described is the lsb of the gain + value. + + For instance assuming 2 gpios + gain-gpios = <&gpio1 GPIO_ACTIVE_HIGH> <&gpio2 GPIO_ACTIVE_HIGH>; + The gain value will be the following: + + gpio1 | gpio2 | gain + ------+-------+----- + 0 | 0 | 0b00 -> 0 + 1 | 0 | 0b01 -> 1 + 0 | 1 | 0b10 -> 2 + 1 | 1 | 0b11 -> 3 + ------+-------+----- + + Note: The gain value, bits set to 1 or 0, indicate the state active (bit + set) or the state inactive (bit unset) of the related GPIO. The + physical voltage corresponding to this active/inactive state is + given by the GPIO_ACTIVE_HIGH and GPIO_ACTIVE_LOW flags. + + minItems: 1 + maxItems: 16 + + gain-ranges: + $ref: /schemas/types.yaml#/definitions/int32-matrix + description: | + A list of one or more ranges of possible values. Each range is defined by + the first and last point in the range. Each point is defined by the pair + (GPIOs value, Gain in 0.01 dB unit). + + Ranges can be contiguous or holes can be present between ranges if some + gpios value should not be used. Also in a range the first point and the + last point can be identical. In that case, the range contains only one + item, the given point. + + items: + items: + - description: GPIOs value of the first point in the range + - description: Gain in 0.01 dB unit of the first point in the range + - description: GPIOs value of the last point in the range + - description: Gain in 0.01 dB unit of the last point in the range + description: | + A range defines a linear function (linear in dB) from the first point + to the last point, both included. The number of items in the range is + N = abs(first_point.gpio_value - last_point.gpio_value) + 1 + + It allows to define the gain range from the first_point.gain to + the last_point.gain, both points included. + + Gain (0.01 dB unit) + ^ + | last + +- - - - - - - - - - + point + | + . + | + . + | + . + +- - - - + . + | first . . + | point . . + | . . + +--------+-----------+---> gpios + value + + Note: Even if first_point.gpio_value is lower than last_point.gpio_value + and first_point.gain is lower than last_point.gain in the above + graphic, all combination of values are supported leading to an + increasing or a decreasing linear segment. + + minItems: 1 + maxItems: 65536 + + gain-labels: + $ref: /schemas/types.yaml#/definitions/string-array + minItems: 2 + maxItems: 65536 + description: | + List of the gain labels attached to the combination of GPIOs controlling + the gain. The first label is related to the gain value 0, the second label + is related to the gain value 1 and so on. + + With 2 GPIOs controlling the gain, GPIOs value can be 0, 1, 2 and 3. + Assuming that gain value set the hardware according to the following + table: + + GPIOs | Hardware + value | amplification + ------+-------------- + 0 | Low + 1 | Middle + 2 | High + 3 | Max + ------+-------------- + + The description using gain labels can be: + gain-labels = "Low", "Middle", "High", "Max"; + +dependencies: + gain-ranges: [ gain-gpios ] + gain-labels: [ gain-gpios ] + +required: + - compatible + - vdd-supply + +anyOf: + - required: + - gain-gpios + - required: + - mute-gpios + - required: + - bypass-gpios + +allOf: + - $ref: component-common.yaml# + - if: + required: + - gain-ranges + then: + properties: + gain-labels: false + - if: + required: + - gain-labels + then: + properties: + gain-ranges: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + /* Gain controlled by gpios */ + amplifier-0 { + compatible = "gpio-audio-amp-mono"; + vdd-supply = <®ulator>; + gain-gpios = <&gpio 0 GPIO_ACTIVE_HIGH>, <&gpio 1 GPIO_ACTIVE_HIGH>; + }; + + /* Gain controlled by gpio using a simple range on a stereo amplifier */ + amplifier-1 { + compatible = "gpio-audio-amp-stereo"; + vdd-supply = <®ulator>; + gain-gpios = <&gpio 0 GPIO_ACTIVE_HIGH>, <&gpio 1 GPIO_ACTIVE_HIGH>; + gain-ranges = <0 (-300) 3 600>; + }; + + /* Gain controlled by gpio with labels */ + amplifier-3 { + compatible = "gpio-audio-amp-mono"; + vdd-supply = <®ulator>; + gain-gpios = <&gpio 0 GPIO_ACTIVE_HIGH>; + gain-labels = "Low", "High"; + }; + + /* A mutable stereo amplifier without any gain control */ + amplifier-4 { + compatible = "gpio-audio-amp-stereo"; + vdd-supply = <®ulator>; + mute-gpios = <&gpio 0 GPIO_ACTIVE_HIGH>; + }; + + /* + * Several supplies, gain controlled using more complex ranges, mute and + * bypass. + * + * Assuming 3 gpios for controlling the gain with the following table + * gpios value Gain + * 0b000 Do not use (gpios value not allowed) + * 0b001 - 3dB + * 0b010 + 3dB + * 0b011 + 10dB + * 0b100 Do not use (gpios value not allowed) + * 0b101 + 6dB + * 0b110 + 7dB + * 0b111 + 8dB + */ + amplifier-5 { + compatible = "gpio-audio-amp-mono"; + vdd-supply = <®ulator>; + vddio-supply = <®ulator1>; + vdda1-supply = <®ulator2>; + gain-gpios = <&gpio 0 GPIO_ACTIVE_HIGH>, + <&gpio 1 GPIO_ACTIVE_HIGH>, + <&gpio 2 GPIO_ACTIVE_HIGH>; + gain-ranges = <1 (-300) 2 300>, + <3 1000 3 1000>, + <5 600 7 800>; + mute-gpios = <&gpio 3 GPIO_ACTIVE_HIGH>; + bypass-gpios = <&gpio 4 GPIO_ACTIVE_HIGH>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.yaml b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.yaml new file mode 100644 index 000000000..5171f9846 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.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/hisilicon,hi6210-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: HiSilicon hi6210 I2S controller + +maintainers: + - John Stultz <john.stultz@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: hisilicon,hi6210-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 2 + + clock-names: + items: + - const: dacodec + - const: i2s-base + + dmas: + maxItems: 2 + + dma-names: + items: + - const: tx + - const: rx + + hisilicon,sysctrl-syscon: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle to sysctrl syscon + + "#sound-dai-cells": + const: 1 + description: | + The dai cell indexes reference the following interfaces: + 0: S2 interface + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - hisilicon,sysctrl-syscon + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/hi6220-clock.h> + + i2s@f7118000 { + compatible = "hisilicon,hi6210-i2s"; + reg = <0xf7118000 0x8000>; + interrupts = <GIC_SPI 123 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&sys_ctrl HI6220_DACODEC_PCLK>, + <&sys_ctrl HI6220_BBPPLL0_DIV>; + clock-names = "dacodec", "i2s-base"; + dmas = <&dma0 14>, <&dma0 15>; + dma-names = "tx", "rx"; + hisilicon,sysctrl-syscon = <&sys_ctrl>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/img,i2s-in.txt b/Documentation/devicetree/bindings/sound/img,i2s-in.txt new file mode 100644 index 000000000..423265cfc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,i2s-in.txt @@ -0,0 +1,47 @@ +Imagination Technologies I2S Input Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,i2s-in" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Must include the following entry: + "sys" The system clock + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "rx" Single DMA channel used by all active I2S channels + + - img,i2s-channels : Number of I2S channels instantiated in the I2S in block + +Optional Properties: + + - interrupts : Contains the I2S in interrupts. Depending on + the configuration, there may be no interrupts, one interrupt, + or an interrupt per I2S channel. For the case where there is + one interrupt per channel, the interrupts should be listed + in ascending channel order + + - resets: Contains a phandle to the I2S in reset signal + + - reset-names: Contains the reset signal name "rst" + +Example: + +i2s_in: i2s-in@18100800 { + compatible = "img,i2s-in"; + reg = <0x18100800 0x200>; + interrupts = <GIC_SHARED 7 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 30 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_I2S_IN>; + clock-names = "sys"; + img,i2s-channels = <6>; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,i2s-out.txt b/Documentation/devicetree/bindings/sound/img,i2s-out.txt new file mode 100644 index 000000000..6b0ee9b7e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,i2s-out.txt @@ -0,0 +1,51 @@ +Imagination Technologies I2S Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,i2s-out" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Must include the following entries: + "sys" The system clock + "ref" The reference clock + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" Single DMA channel used by all active I2S channels + + - img,i2s-channels : Number of I2S channels instantiated in the I2S out block + + - resets: Contains a phandle to the I2S out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the I2S out interrupts. Depending on + the configuration, there may be no interrupts, one interrupt, + or an interrupt per I2S channel. For the case where there is + one interrupt per channel, the interrupts should be listed + in ascending channel order + +Example: + +i2s_out: i2s-out@18100a00 { + compatible = "img,i2s-out"; + reg = <0x18100A00 0x200>; + interrupts = <GIC_SHARED 13 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 23 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_I2S_OUT>, + <&clk_core CLK_I2S>; + clock-names = "sys", "ref"; + img,i2s-channels = <6>; + resets = <&pistachio_reset PISTACHIO_RESET_I2S_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,parallel-out.txt b/Documentation/devicetree/bindings/sound/img,parallel-out.txt new file mode 100644 index 000000000..37a3f94cc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,parallel-out.txt @@ -0,0 +1,44 @@ +Imagination Technologies Parallel Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,parallel-out". + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device. + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "sys" The system clock + "ref" The reference clock + + - resets: Contains a phandle to the parallel out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the parallel out interrupt, if present + +Example: + +parallel_out: parallel-out@18100c00 { + compatible = "img,parallel-out"; + reg = <0x18100C00 0x100>; + interrupts = <GIC_SHARED 19 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 16 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_PAUD_OUT>, + <&clk_core CLK_AUDIO_DAC>; + clock-names = "sys", "ref"; + resets = <&pistachio_reset PISTACHIO_RESET_PRL_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt b/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt new file mode 100644 index 000000000..4cc18fc04 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt @@ -0,0 +1,18 @@ +Pistachio internal DAC DT bindings + +Required properties: + + - compatible: "img,pistachio-internal-dac" + + - img,cr-top : Must contain a phandle to the top level control syscon + node which contains the internal dac control registers + + - VDD-supply : Digital power supply regulator (+1.8V or +3.3V) + +Examples: + +internal_dac: internal-dac { + compatible = "img,pistachio-internal-dac"; + img,cr-top = <&cr_top>; + VDD-supply = <&supply3v3>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.txt b/Documentation/devicetree/bindings/sound/img,spdif-in.txt new file mode 100644 index 000000000..f7ea8c87b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,spdif-in.txt @@ -0,0 +1,41 @@ +Imagination Technologies SPDIF Input Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,spdif-in" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "rx" + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Includes the following entries: + "sys" The system clock + +Optional Properties: + + - resets: Should contain a phandle to the spdif in reset signal, if any + + - reset-names: Should contain the reset signal name "rst", if a + reset phandle is given + + - interrupts : Contains the spdif in interrupt, if present + +Example: + +spdif_in: spdif-in@18100e00 { + compatible = "img,spdif-in"; + reg = <0x18100E00 0x100>; + interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 15 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_SPDIF_IN>; + clock-names = "sys"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/img,spdif-out.txt b/Documentation/devicetree/bindings/sound/img,spdif-out.txt new file mode 100644 index 000000000..413ed8b01 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,spdif-out.txt @@ -0,0 +1,44 @@ +Imagination Technologies SPDIF Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,spdif-out" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "sys" The system clock + "ref" The reference clock + + - resets: Contains a phandle to the spdif out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the parallel out interrupt, if present + +Example: + +spdif_out: spdif-out@18100d00 { + compatible = "img,spdif-out"; + reg = <0x18100D00 0x100>; + interrupts = <GIC_SHARED 21 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 14 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_SPDIF_OUT>, + <&clk_core CLK_SPDIF>; + clock-names = "sys", "ref"; + resets = <&pistachio_reset PISTACHIO_RESET_SPDIF_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml new file mode 100644 index 000000000..950e3eab2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -0,0 +1,134 @@ +# 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 + $ref: tdm-slot.yaml# + + 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 + - left_j + - right_j + - dsp_a + - dsp_b + - pdm + - msb + - lsb + + dai-tdm-slot-num: true + + dai-tdm-slot-width: true + + playback-only: + description: link is used only for playback + $ref: /schemas/types.yaml#/definitions/flag + + capture-only: + description: link is used only for capture + $ref: /schemas/types.yaml#/definitions/flag + + 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 + $ref: /schemas/types.yaml#/definitions/flag + + required: + - link-name + - cpu + + allOf: + - not: + required: + - playback-only + - capture-only + + additionalProperties: false + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + sound-ak4458 { + compatible = "fsl,imx-audio-card"; + model = "ak4458-audio"; + pri-dai-link { + link-name = "akcodec"; + format = "i2s"; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + fe-dai-link { + link-name = "HiFi-ASRC-FE"; + format = "i2s"; + cpu { + sound-dai = <&easrc>; + }; + }; + be-dai-link { + link-name = "HiFi-ASRC-BE"; + format = "dsp_b"; + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000..e7e7bb65c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-hdmi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio complex with HDMI + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-hdmi + - fsl,imx-audio-sii902x + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + + hdmi-out: + type: boolean + description: | + This is a boolean property. If present, the transmitting function + of HDMI will be enabled, indicating there's a physical HDMI out + connector or jack on the board or it's connecting to some other IP + block, such as an HDMI encoder or display-controller. + + hdmi-in: + type: boolean + description: | + This is a boolean property. If present, the receiving function of + HDMI will be enabled, indicating there is a physical HDMI in + connector/jack on the board. + +required: + - compatible + - model + - audio-cpu + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "fsl,imx-audio-hdmi"; + model = "audio-hdmi"; + audio-cpu = <&aud2htx>; + hdmi-out; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000..dab45c310 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel@pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml b/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml new file mode 100644 index 000000000..5e11ce2c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/infineon,peb2466.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Infineon PEB2466 codec + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The Infineon PEB2466 codec is a programmable DSP-based four channels codec + with filters capabilities. + + The time-slots used by the codec must be set and so, the properties + 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and + 'dai-tdm-slot-rx-mask' must be present in the sound card node for sub-nodes + that involve the codec. The codec uses one 8bit time-slot per channel. + 'dai-tdm-tdm-slot-with' must be set to 8. + + The PEB2466 codec also supports 28 gpios (signaling pins). + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml + - $ref: dai-common.yaml# + +properties: + compatible: + const: infineon,peb2466 + + reg: + description: + SPI device address. + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + spi-max-frequency: + maximum: 8192000 + + reset-gpios: + description: + GPIO used to reset the device. + maxItems: 1 + + firmware-name: + maxItems: 1 + description: + Filters coefficients file to load. If this property is omitted, internal + filters are disabled. + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - '#sound-dai-cells' + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + compatible = "infineon,peb2466"; + reg = <0>; + spi-max-frequency = <8192000>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,aic.yaml b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml new file mode 100644 index 000000000..d15c000f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $nodename: + pattern: '^audio-controller@' + + compatible: + oneOf: + - enum: + - ingenic,jz4740-i2s + - ingenic,jz4760-i2s + - ingenic,jz4770-i2s + - ingenic,jz4780-i2s + - ingenic,x1000-i2s + - items: + - const: ingenic,jz4725b-i2s + - const: ingenic,jz4740-i2s + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: AIC clock + - description: I2S clock + + clock-names: + items: + - const: aic + - const: i2s + + dmas: + items: + - description: DMA controller phandle and request line for I2S RX + - description: DMA controller phandle and request line for I2S TX + + dma-names: + items: + - const: rx + - const: tx + +unevaluatedProperties: false + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> + aic: audio-controller@10020000 { + compatible = "ingenic,jz4740-i2s"; + reg = <0x10020000 0x38>; + + #sound-dai-cells = <0>; + + interrupt-parent = <&intc>; + interrupts = <18>; + + clocks = <&cgu JZ4740_CLK_AIC>, + <&cgu JZ4740_CLK_I2S>; + clock-names = "aic", "i2s"; + + dmas = <&dmac 25 0xffffffff>, <&dmac 24 0xffffffff>; + dma-names = "rx", "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml new file mode 100644 index 000000000..b58b90850 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $nodename: + pattern: '^audio-codec@.*' + + compatible: + oneOf: + - enum: + - ingenic,jz4770-codec + - ingenic,jz4760-codec + - ingenic,jz4725b-codec + - ingenic,jz4740-codec + - items: + - const: ingenic,jz4760b-codec + - const: ingenic,jz4760-codec + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: aic + + '#sound-dai-cells': + const: 0 + +unevaluatedProperties: false + +required: + - compatible + - reg + - clocks + - clock-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/ingenic,jz4740-cgu.h> + codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + #sound-dai-cells = <0>; + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml new file mode 100644 index 000000000..dca617860 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright 2020 Intel Corporation +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Intel KeemBay I2S + +maintainers: + - Daniele Alessandrelli <daniele.alessandrelli@intel.com> + - Paul J. Murphy <paul.j.murphy@intel.com> + +description: | + Intel KeemBay I2S + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - intel,keembay-i2s + - intel,keembay-tdm + - intel,keembay-hdmi-i2s + + "#sound-dai-cells": + const: 0 + + reg: + items: + - description: I2S registers + - description: I2S gen configuration + + reg-names: + items: + - const: i2s-regs + - const: i2s_gen_cfg + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: osc + - const: apb_clk + + dmas: + items: + - description: DMA TX channel + - description: DMA RX channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #define KEEM_BAY_PSS_AUX_I2S3 + #define KEEM_BAY_PSS_I2S3 + i2s@20140000 { + compatible = "intel,keembay-i2s"; + #sound-dai-cells = <0>; + reg = <0x20140000 0x200>, /* I2S registers */ + <0x202a00a4 0x4>; /* I2S gen configuration */ + reg-names = "i2s-regs", "i2s_gen_cfg"; + interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; + clock-names = "osc", "apb_clk"; + clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>; + dmas = <&axi_dma0 29>, <&axi_dma0 33>; + dma-names = "tx", "rx"; + }; diff --git a/Documentation/devicetree/bindings/sound/invensense,ics43432.yaml b/Documentation/devicetree/bindings/sound/invensense,ics43432.yaml new file mode 100644 index 000000000..7bd984817 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/invensense,ics43432.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/invensense,ics43432.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Invensense ICS-43432-compatible MEMS Microphone with I2S Output + +maintainers: + - Oleksij Rempel <o.rempel@pengutronix.de> + +description: + The ICS-43432 and compatible MEMS microphones output audio over an I2S + interface and require no software configuration. The only host connection + is the I2S bus. The microphone requires an I2S clock frequency between + 460 kHz and 3.379 MHz and 64 clock cycles per stereo frame. Each frame + contains 32-bit slots per channel, with 24 bits carrying audio data. + A hardware pin determines whether the microphone outputs audio on the + left or right channel of the I2S frame. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - invensense,ics43432 + - cui,cmm-4030d-261 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + ics43432: ics43432 { + compatible = "invensense,ics43432"; + + port { + endpoint { + remote-endpoint = <&i2s1_endpoint>; + dai-format = "i2s"; + }; + }; + + }; diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml new file mode 100644 index 000000000..b36c35e5d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1303.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1303 Audio Amplifier + +maintainers: + - Kiseok Jo <kiseok.jo@irondevice.com> + +description: + SMA1303 digital class-D audio amplifier + with an integrated boost converter. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1303 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1303"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml new file mode 100644 index 000000000..1e2a038d0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml @@ -0,0 +1,53 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1307.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1307 Audio Amplifier + +maintainers: + - Kiseok Jo <kiseok.jo@irondevice.com> + +description: + SMA1307 boosted digital speaker amplifier with feedback-loop. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1307a + - irondevice,sma1307aq + description: + If a 'q' is added, it indicated the product is AEC-Q100 + qualified for automotive applications. SMA1307A supports + both WLCSP and QFN packages. However, SMA1307AQ only + supports the QFN package. + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1307a"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml new file mode 100644 index 000000000..a67b79cbe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bluetooth SCO Audio Codec + +maintainers: + - Mark Brown <broonie@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + '#sound-dai-cells': + enum: + - 0 + + # For Wideband PCM + - 1 + + compatible: + enum: + - delta,dfbmcs320 + - linux,bt-sco + +required: + - '#sound-dai-cells' + - compatible + +unevaluatedProperties: false + +examples: + - | + codec { + #sound-dai-cells = <0>; + compatible = "linux,bt-sco"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/linux,spdif.yaml b/Documentation/devicetree/bindings/sound/linux,spdif.yaml new file mode 100644 index 000000000..aea6230db --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,spdif.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dummy SPDIF Transmitter/Receiver + +maintainers: + - Mark Brown <broonie@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - linux,spdif-dit + - linux,spdif-dir + + "#sound-dai-cells": + const: 0 + + sound-name-prefix: true + + port: + $ref: /schemas/graph.yaml#/properties/port + +required: + - "#sound-dai-cells" + - compatible + +additionalProperties: false + +examples: + - | + spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000..dc7f4afbb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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> + - Binbin Zhou <zhoubinbin@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. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - loongson,ls-audio-card # Loongson-2K1000/Loongson-2K2000/LS7A + - loongson,ls2k0300-forever-pi-audio-card # CTCISZ Forever Pi + - loongson,ls2k0300-dl2k0300b-audio-card # ATK-DL2K0300B + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + hp-ctl-gpios: + maxItems: 1 + description: The GPIO that control the headphones + + hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + +required: + - compatible + - mclk-fs + - cpu + - codec + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "loongson,ls2k0300-dl2k0300b-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + hp-det-gpios = <&gpio 81 GPIO_ACTIVE_HIGH>; + spkr-en-gpios = <&gpio 86 GPIO_ACTIVE_HIGH>; + hp-ctl-gpios = <&gpio 87 GPIO_ACTIVE_HIGH>; + audio-routing = + "Headphone", "LOUT1", + "Headphone", "ROUT1", + "Speaker", "LOUT2", + "Speaker", "ROUT2"; + + cpu { + sound-dai = <&i2s>; + }; + + codec { + sound-dai = <&es8388>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/loongson,ls1b-ac97.yaml b/Documentation/devicetree/bindings/sound/loongson,ls1b-ac97.yaml new file mode 100644 index 000000000..1c6a2771f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls1b-ac97.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls1b-ac97.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson-1 AC97 Controller + +maintainers: + - Keguang Zhang <keguang.zhang@gmail.com> + +description: + The Loongson-1 AC97 controller supports 2-channel stereo output and input. + It is paired with the DMA engine to handle playback and capture functions. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - const: loongson,ls1b-ac97 + - items: + - enum: + - loongson,ls1a-ac97 + - loongson,ls1c-ac97 + - const: loongson,ls1b-ac97 + + reg: + maxItems: 3 + + reg-names: + items: + - const: ac97 + - const: audio-tx + - const: audio-rx + + dmas: + maxItems: 2 + + dma-names: + items: + - const: tx + - const: rx + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + audio-controller@1fe74000 { + compatible = "loongson,ls1b-ac97"; + reg = <0x1fe74000 0x60>, <0x1fe72420 0x4>, <0x1fe74c4c 0x4>; + reg-names = "ac97", "audio-tx", "audio-rx"; + dmas = <&dma 1>, <&dma 2>; + dma-names = "tx", "rx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/loongson,ls2k1000-i2s.yaml b/Documentation/devicetree/bindings/sound/loongson,ls2k1000-i2s.yaml new file mode 100644 index 000000000..51e23c189 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls2k1000-i2s.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls2k1000-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson-2K1000 I2S controller + +maintainers: + - Binbin Zhou <zhoubinbin@loongson.cn> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - loongson,ls2k0300-i2s + - loongson,ls2k1000-i2s + + reg: + minItems: 1 + items: + - description: Loongson I2S controller Registers. + - description: APB DMA config register for Loongson I2S controller. + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + dmas: + maxItems: 2 + + dma-names: + items: + - const: tx + - const: rx + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - dmas + - dma-names + - '#sound-dai-cells' + +unevaluatedProperties: false + +if: + properties: + compatible: + contains: + enum: + - loongson,ls2k1000-i2s + +then: + properties: + reg: + minItems: 2 + +else: + properties: + reg: + maxItems: 1 + +examples: + - | + #include <dt-bindings/clock/loongson,ls2k-clk.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2s@1fe2d000 { + compatible = "loongson,ls2k1000-i2s"; + reg = <0x1fe2d000 0x14>, + <0x1fe00438 0x8>; + interrupt-parent = <&liointc0>; + interrupts = <5 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk LOONGSON2_APB_CLK>; + dmas = <&apbdma2 0>, <&apbdma3 0>; + dma-names = "tx", "rx"; + #sound-dai-cells = <0>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml new file mode 100644 index 000000000..4193d17d1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$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/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt new file mode 100644 index 000000000..2ea85d5be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt @@ -0,0 +1,27 @@ +Marvell PXA2xx audio complex + +This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series. + +Required properties: + - compatible: should be one of the following: + "marvell,pxa250-ac97" + "marvell,pxa270-ac97" + "marvell,pxa300-ac97" + - reg: device MMIO address space + - interrupts: single interrupt generated by AC97 IP + - clocks: input clock of the AC97 IP, refer to clock-bindings.txt + +Optional properties: + - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt + - reset-gpios: gpio used for AC97 reset, refer to gpio.txt + +Example: + ac97: sound@40500000 { + compatible = "marvell,pxa250-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pmux_ac97_default >; + }; diff --git a/Documentation/devicetree/bindings/sound/max98373.txt b/Documentation/devicetree/bindings/sound/max98373.txt new file mode 100644 index 000000000..456cb1c59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98373.txt @@ -0,0 +1,40 @@ +Maxim Integrated MAX98373 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98373" + + - reg : the I2C address of the device. + +Optional properties: + + - maxim,vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + slot range : 0 ~ 15, Default : 0 + + - maxim,imon-slot-no : slot number used to send current information + slot range : 0 ~ 15, Default : 0 + + - maxim,spkfb-slot-no : slot number used to send speaker feedback information + slot range : 0 ~ 15, Default : 0 + + - maxim,interleave-mode : For cases where a single combined channel + for the I/V sense data is not sufficient, the device can also be configured + to share a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame interleaved + on a single output channel. + Boolean, define to enable the interleave mode, Default : false + +Example: + +codec: max98373@31 { + compatible = "maxim,max98373"; + reg = <0x31>; + maxim,vmon-slot-no = <0>; + maxim,imon-slot-no = <1>; + maxim,spkfb-slot-no = <2>; + maxim,interleave-mode; +}; diff --git a/Documentation/devicetree/bindings/sound/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt new file mode 100644 index 000000000..e0d4e95e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9860.txt @@ -0,0 +1,28 @@ +MAX9860 Mono Audio Voice Codec + +Required properties: + + - compatible : "maxim,max9860" + + - reg : the I2C address of the device + + - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for + the device, as covered in bindings/regulator/regulator.txt + + - clock-names : Required element: "mclk". + + - clocks : A clock specifier for the clock connected as MCLK. + +Examples: + + max9860: max9860@10 { + compatible = "maxim,max9860"; + reg = <0x10>; + + AVDD-supply = <®_1v8>; + DVDD-supply = <®_1v8>; + DVDDIO-supply = <®_3v0>; + + clock-names = "mclk"; + clocks = <&pck2>; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max9759.yaml b/Documentation/devicetree/bindings/sound/maxim,max9759.yaml new file mode 100644 index 000000000..a76ee6a63 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9759.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max9759.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim MAX9759 Speaker Amplifier + +maintainers: + - Otabek Nazrullaev <otabeknazrullaev1998@gmail.com> + +properties: + compatible: + const: maxim,max9759 + + shutdown-gpios: + maxItems: 1 + description: the gpio connected to the shutdown pin + + mute-gpios: + maxItems: 1 + description: the gpio connected to the mute pin + + gain-gpios: + maxItems: 2 + description: the 2 gpios connected to the g1 and g2 pins + +required: + - compatible + - shutdown-gpios + - mute-gpios + - gain-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + amplifier { + compatible = "maxim,max9759"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>; + gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>, + <&gpio3 25 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98088.yaml b/Documentation/devicetree/bindings/sound/maxim,max98088.yaml new file mode 100644 index 000000000..e4a2967e1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98088.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98088.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MAX98088 audio CODEC + +maintainers: + - Abdulrasaq Lawani <abdulrasaqolawani@gmail.com> + +properties: + compatible: + enum: + - maxim,max98088 + - maxim,max98089 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@10 { + compatible = "maxim,max98089"; + reg = <0x10>; + clocks = <&clks 0>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98090.yaml b/Documentation/devicetree/bindings/sound/maxim,max98090.yaml new file mode 100644 index 000000000..9df1296aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98090.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98090/MAX98091 audio codecs + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + +description: | + Pins on the device (for linking into audio routes): + MIC1, MIC2, DMICL, DMICR, IN1, IN2, IN3, IN4, IN5, IN6, IN12, IN34, IN56, + HPL, HPR, SPKL, SPKR, RCVL, RCVR, MICBIAS + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98090 + - maxim,max98091 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + interrupts: + maxItems: 1 + + maxim,dmic-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 2500000 + description: + DMIC clock frequency + + maxim,micbias: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1, 2, 3 ] + default: 3 + description: | + Micbias voltage applied to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 IRQ_TYPE_EDGE_FALLING>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98095.yaml b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml new file mode 100644 index 000000000..76ea4fe71 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98095.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98095 audio codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98095 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98357a.yaml b/Documentation/devicetree/bindings/sound/maxim,max98357a.yaml new file mode 100644 index 000000000..83ba8666f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98357a.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98357a.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98357A/MAX98360A amplifier + +maintainers: + - Tzung-Bi Shih <tzungbi@kernel.org> + +description: + Maxim Integrated MAX98357A/MAX98360A is a digital pulse-code modulation (PCM) + input Class D amplifier. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98357a + - maxim,max98360a + + '#sound-dai-cells': + const: 0 + + sdmode-gpios: + maxItems: 1 + description: + Chip's SD_MODE pin. If missing the chip is always on. + + sdmode-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Delay time for SD_MODE pin changes intended to make I2S clocks ready + before SD_MODE is unmuted in order to avoid the speaker pop noise. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + amplifier { + compatible = "maxim,max98360a"; + #sound-dai-cells = <0>; + sdmode-gpios = <&qcom_pinmux 25 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98371.yaml b/Documentation/devicetree/bindings/sound/maxim,max98371.yaml new file mode 100644 index 000000000..14fba34ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98371.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98371.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim MAX98371 audio codec + +maintainers: + - anish kumar <yesanishhere@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: maxim,max98371 + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@31 { + compatible = "maxim,max98371"; + reg = <0x31>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000..d35dd8408 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,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> + +allOf: + - $ref: dai-common.yaml# + +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. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 8388607 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + reset-gpios = <&gpio 9 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98504.yaml b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml new file mode 100644 index 000000000..6d33bb4a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98504.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98504 class D mono speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + +description: + Maxim Integrated MAX98504 speaker amplifier supports I2C control interface + with an IRQ output signal, PCM and PDM digital audio interface (DAI) and a + differential analog input. + +properties: + compatible: + const: maxim,max98504 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + DIOVDD-supply: true + DVDD-supply: true + PVDD-supply: true + + maxim,brownout-threshold: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 21 + default: 0 + description: + PVDD brownout threshold, where values correspond to 2.6V, 2.65V...3.65V + voltage range. Property also enables the PVDD brownout protection. + + maxim,brownout-attenuation: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 6 + default: 0 + description: + Brownout attenuation to the speaker gain applied during the "attack hold" + and "timed hold" phase, the value must be from 0...6 (dB) range. + + maxim,brownout-attack-hold-ms: + maximum: 255 + default: 0 + description: + Brownout attack hold phase time in ms, VBATBROWN_ATTK_HOLD, register 0x0018. + + maxim,brownout-timed-hold-ms: + maximum: 255 + default: 0 + description: + Brownout timed hold phase time in ms, VBATBROWN_TIME_HOLD, register 0x0019. + + maxim,brownout-release-rate-ms: + maximum: 255 + default: 0 + description: + Brownout release phase step time in ms, VBATBROWN_RELEASE, register 0x001A. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DIOVDD-supply = <&ldo3_reg>; + DVDD-supply = <&ldo3_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000..3f88c7d61 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song@maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max9867.yaml b/Documentation/devicetree/bindings/sound/maxim,max9867.yaml new file mode 100644 index 000000000..0b9a84d33 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9867.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max9867.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX9867 CODEC + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * LOUT + * ROUT + * LINL + * LINR + * MICL + * MICR + * DMICL + * DMICR + +maintainers: + - Ladislav Michl <ladis@linux-mips.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max9867 + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@18 { + compatible = "maxim,max9867"; + #sound-dai-cells = <0>; + reg = <0x18>; + clocks = <&codec_clk>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000..121e8d2d4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee@maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml new file mode 100644 index 000000000..30f331366 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml @@ -0,0 +1,139 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt2701-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End (AFE) PCM controller for mt2701 + +description: + The AFE PCM node must be a subnode of the MediaTek audsys device tree node. + +maintainers: + - Eugen Hristev <eugen.hristev@collabora.com> + +properties: + compatible: + enum: + - mediatek,mt2701-audio + - mediatek,mt7622-audio + + interrupts: + items: + - description: AFE interrupt + - description: ASYS interrupt + + interrupt-names: + items: + - const: afe + - const: asys + + power-domains: + maxItems: 1 + + clocks: + minItems: 34 + items: + - description: audio infra sys clock + - description: top audio mux 1 + - description: top audio mux 2 + - description: top audio sys a1 clock + - description: top audio sys a2 clock + - description: i2s0 source selection + - description: i2s1 source selection + - description: i2s2 source selection + - description: i2s3 source selection + - description: i2s0 source divider + - description: i2s1 source divider + - description: i2s2 source divider + - description: i2s3 source divider + - description: i2s0 master clock + - description: i2s1 master clock + - description: i2s2 master clock + - description: i2s3 master clock + - description: i2so0 hopping clock + - description: i2so1 hopping clock + - description: i2so2 hopping clock + - description: i2so3 hopping clock + - description: i2si0 hopping clock + - description: i2si1 hopping clock + - description: i2si2 hopping clock + - description: i2si3 hopping clock + - description: asrc0 output clock + - description: asrc1 output clock + - description: asrc2 output clock + - description: asrc3 output clock + - description: audio front end pd clock + - description: audio front end conn pd clock + - description: top audio a1 sys pd + - description: top audio a2 sys pd + - description: audio merge interface pd + - description: HADDS2 PLL 294 MHz (HDMI audio path root) + - description: HDMI audio interface pd + - description: S/PDIF interface pd + - description: audio APLL root pd + + clock-names: + minItems: 34 + items: + - const: infra_sys_audio_clk + - const: top_audio_mux1_sel + - const: top_audio_mux2_sel + - const: top_audio_a1sys_hp + - const: top_audio_a2sys_hp + - const: i2s0_src_sel + - const: i2s1_src_sel + - const: i2s2_src_sel + - const: i2s3_src_sel + - const: i2s0_src_div + - const: i2s1_src_div + - const: i2s2_src_div + - const: i2s3_src_div + - const: i2s0_mclk_en + - const: i2s1_mclk_en + - const: i2s2_mclk_en + - const: i2s3_mclk_en + - const: i2so0_hop_ck + - const: i2so1_hop_ck + - const: i2so2_hop_ck + - const: i2so3_hop_ck + - const: i2si0_hop_ck + - const: i2si1_hop_ck + - const: i2si2_hop_ck + - const: i2si3_hop_ck + - const: asrc0_out_ck + - const: asrc1_out_ck + - const: asrc2_out_ck + - const: asrc3_out_ck + - const: audio_afe_pd + - const: audio_afe_conn_pd + - const: audio_a1sys_pd + - const: audio_a2sys_pd + - const: audio_mrgif_pd + - const: hadds2pll_294m + - const: audio_hdmi_pd + - const: audio_spdf_pd + - const: audio_apll_pd + +required: + - compatible + - interrupts + - interrupt-names + - power-domains + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + const: mediatek,mt7622-audio + then: + properties: + clocks: + maxItems: 34 + clock-names: + maxItems: 34 + +additionalProperties: false diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt2701-hdmi-audio.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-hdmi-audio.yaml new file mode 100644 index 000000000..9d5a8166e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-hdmi-audio.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/mediatek,mt2701-hdmi-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT2701/MT7623N HDMI audio output + +maintainers: + - Daniel Golle <daniel@makrotopia.org> + +description: + Sound card routing the MT2701/MT7623N Audio Front End HDMI + playback path to the on-chip HDMI transmitter. The AFE + provides the DMA memif and internal I2S engine; the HDMI + transmitter acts as the audio codec on the serialised link. + +properties: + compatible: + oneOf: + - const: mediatek,mt2701-hdmi-audio + - items: + - const: mediatek,mt7623n-hdmi-audio + - const: mediatek,mt2701-hdmi-audio + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle of the MT2701/MT7623N AFE platform node. + + mediatek,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle of the HDMI transmitter acting as audio codec. + +required: + - compatible + - mediatek,platform + - mediatek,audio-codec + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "mediatek,mt7623n-hdmi-audio", + "mediatek,mt2701-hdmi-audio"; + mediatek,platform = <&afe>; + mediatek,audio-codec = <&hdmi0>; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml new file mode 100644 index 000000000..bb6a405b2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt2701-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT2701 with WM8960 CODEC + +maintainers: + - Kartik Agarwala <agarwala.kartik@gmail.com> + +properties: + compatible: + const: mediatek,mt2701-wm8960-machine + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT2701 ASoC platform. + + 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. + + mediatek,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8960 audio codec. + +required: + - compatible + - mediatek,platform + - audio-routing + - mediatek,audio-codec + - pinctrl-names + - pinctrl-0 + +additionalProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt2701-wm8960-machine"; + mediatek,platform = <&afe>; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + mediatek,audio-codec = <&wm8960>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt6351-sound.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt6351-sound.yaml new file mode 100644 index 000000000..b422e238b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt6351-sound.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt6351-sound.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT6351 Audio CODEC + +maintainers: + - KaiChieh Chuang <kaichieh.chuang@mediatek.com> + +description: + MT6351 Audio CODEC is a part of the MediaTek MT6351 PMIC. + It communicates with the SoC through the MediaTek PMIC wrapper(pwrap). + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: mediatek,mt6351-sound + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt6351-sound"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml new file mode 100644 index 000000000..398efdfe0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml @@ -0,0 +1,160 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt7986-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AFE PCM controller for MT7986 + +maintainers: + - Maso Huang <maso.huang@mediatek.com> + +properties: + compatible: + oneOf: + - const: mediatek,mt7986-afe + - items: + - enum: + - mediatek,mt7981-afe + - mediatek,mt7988-afe + - const: mediatek,mt7986-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 5 + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + - description: audio i2s/pcm mck + + clock-names: + minItems: 5 + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + - const: i2s_m + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + const: mediatek,mt7986-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + + - if: + properties: + compatible: + contains: + const: mediatek,mt7981-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + + - if: + properties: + compatible: + contains: + const: mediatek,mt7988-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + - description: audio i2s/pcm mck + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + - const: i2s_m + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/clock/mt7986-clk.h> + + afe@11210000 { + compatible = "mediatek,mt7986-afe"; + reg = <0x11210000 0x9000>; + interrupts = <GIC_SPI 106 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&infracfg_ao CLK_INFRA_AUD_BUS_CK>, + <&infracfg_ao CLK_INFRA_AUD_26M_CK>, + <&infracfg_ao CLK_INFRA_AUD_L_CK>, + <&infracfg_ao CLK_INFRA_AUD_AUD_CK>, + <&infracfg_ao CLK_INFRA_AUD_EG2_CK>; + clock-names = "bus_ck", + "26m_ck", + "l_ck", + "aud_ck", + "eg2_ck"; + assigned-clocks = <&topckgen CLK_TOP_A1SYS_SEL>, + <&topckgen CLK_TOP_AUD_L_SEL>, + <&topckgen CLK_TOP_A_TUNER_SEL>; + assigned-clock-parents = <&topckgen CLK_TOP_APLL2_D4>, + <&apmixedsys CLK_APMIXED_APLL2>, + <&topckgen CLK_TOP_APLL2_D4>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml new file mode 100644 index 000000000..f21cad4ba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt7986-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT7986 sound card with WM8960 codec + +maintainers: + - Maso Huang <maso.huang@mediatek.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: mediatek,mt7986-wm8960-sound + + platform: + type: object + additionalProperties: false + properties: + sound-dai: + description: The phandle of MT7986 platform. + maxItems: 1 + required: + - sound-dai + + codec: + type: object + additionalProperties: false + properties: + sound-dai: + description: The phandle of wm8960 codec. + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - audio-routing + - platform + - codec + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt7986-wm8960-sound"; + model = "mt7986-wm8960"; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + + platform { + sound-dai = <&afe>; + }; + + codec { + sound-dai = <&wm8960>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml new file mode 100644 index 000000000..d8993b5d4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8173-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for MT8173 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8173-afe-pcm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: audio infra sys clock + - description: audio top mux + - description: audio intbus mux + - description: apll1 clock + - description: apll2 clock + - description: i2s0 mclk mux + - description: i2s1 mclk mux + - description: i2s2 mclk mux + - description: i2s3 mclk mux + - description: i2s3 bclk mux + + clock-names: + items: + - const: infra_sys_audio_clk + - const: top_pdn_audio + - const: top_pdn_aud_intbus + - const: bck0 + - const: bck1 + - const: i2s0_m + - const: i2s1_m + - const: i2s2_m + - const: i2s3_m + - const: i2s3_b + + power-domains: + maxItems: 1 + + memory-region: + description: memory region for audio DMA buffers + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - power-domains + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8173-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8173-power.h> + + mt8173-afe-pcm@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0x11220000 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + power-domains = <&spm MT8173_POWER_DOMAIN_AUDIO>; + clocks = <&infracfg CLK_INFRA_AUDIO>, + <&topckgen CLK_TOP_AUDIO_SEL>, + <&topckgen CLK_TOP_AUD_INTBUS_SEL>, + <&topckgen CLK_TOP_APLL1_DIV0>, + <&topckgen CLK_TOP_APLL2_DIV0>, + <&topckgen CLK_TOP_I2S0_M_SEL>, + <&topckgen CLK_TOP_I2S1_M_SEL>, + <&topckgen CLK_TOP_I2S2_M_SEL>, + <&topckgen CLK_TOP_I2S3_M_SEL>, + <&topckgen CLK_TOP_I2S3_B_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"; + memory-region = <&afe_dma_mem>; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8173-rt5650-rt5514.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8173-rt5650-rt5514.yaml new file mode 100644 index 000000000..becc7a11f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8173-rt5650-rt5514.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/mediatek,mt8173-rt5650-rt5514.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek MT8173 with RT5650 and RT5514 audio codecs + +maintainers: + - Koro Chen <koro.chen@mediatek.com> + +properties: + compatible: + const: mediatek,mt8173-rt5650-rt5514 + + mediatek,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: Phandles of rt5650 and rt5514 codecs + items: + - description: phandle of rt5650 codec + maxItems: 1 + - description: phandle of rt5514 codec + maxItems: 1 + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8173 ASoC platform. + +required: + - compatible + - mediatek,audio-codec + - mediatek,platform + +additionalProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8173-rt5650-rt5514"; + mediatek,audio-codec = <&rt5650>, <&rt5514>; + mediatek,platform = <&afe>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8183-audio.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8183-audio.yaml new file mode 100644 index 000000000..031b0fa7b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8183-audio.yaml @@ -0,0 +1,228 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8183-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8183 + +maintainers: + - Julien Massot <jmassot@collabora.com> + +properties: + compatible: + const: mediatek,mt8183-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + power-domains: + maxItems: 1 + + memory-region: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: ADDA ADC clock + - description: ADDA6 ADC clock + - description: Audio low-jitter 22.5792m clock + - description: Audio low-jitter 24.576m clock + - description: Audio PLL1 tuner clock + - description: Audio PLL2 tuner clock + - description: I2S1 bit clock + - description: I2S2 bit clock + - description: I2S3 bit clock + - description: I2S4 bit clock + - description: Audio Time-Division Multiplexing interface clock + - description: Powerdown Audio test model clock + - description: Audio infra sys clock + - description: Audio infra 26M clock + - description: Mux for audio clock + - description: Mux for audio internal bus clock + - description: Mux main divider by 4 + - description: Primary audio mux + - description: Primary audio PLL + - description: Secondary audio mux + - description: Secondary audio PLL + - description: Primary audio en-generator clock + - description: Primary PLL divider by 4 for IEC + - description: Secondary audio en-generator clock + - description: Secondary PLL divider by 8 for IEC + - description: Mux selector for I2S port 0 + - description: Mux selector for I2S port 1 + - description: Mux selector for I2S port 2 + - description: Mux selector for I2S port 3 + - description: Mux selector for I2S port 4 + - description: Mux selector for I2S port 5 + - description: APLL1 and APLL2 divider for I2S port 0 + - description: APLL1 and APLL2 divider for I2S port 1 + - description: APLL1 and APLL2 divider for I2S port 2 + - description: APLL1 and APLL2 divider for I2S port 3 + - description: APLL1 and APLL2 divider for I2S port 4 + - description: APLL1 and APLL2 divider for IEC + - description: 26MHz clock for audio subsystem + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_adc_clk + - const: aud_adc_adda6_clk + - const: aud_apll22m_clk + - const: aud_apll24m_clk + - const: aud_apll1_tuner_clk + - const: aud_apll2_tuner_clk + - const: aud_i2s1_bclk_sw + - const: aud_i2s2_bclk_sw + - const: aud_i2s3_bclk_sw + - const: aud_i2s4_bclk_sw + - const: aud_tdm_clk + - const: aud_tml_clk + - const: aud_infra_clk + - const: mtkaif_26m_clk + - const: top_mux_audio + - const: top_mux_aud_intbus + - const: top_syspll_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_i2s3_m_sel + - const: top_i2s4_m_sel + - const: top_i2s5_m_sel + - const: top_apll12_div0 + - const: top_apll12_div1 + - const: top_apll12_div2 + - const: top_apll12_div3 + - const: top_apll12_div4 + - const: top_apll12_divb + - const: top_clk26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8183-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8183-power.h> + #include <dt-bindings/reset/mt8183-resets.h> + + audio-controller { + compatible = "mediatek,mt8183-audio"; + interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>; + resets = <&watchdog MT8183_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + power-domains = <&spm MT8183_POWER_DOMAIN_AUDIO>; + clocks = <&audiosys CLK_AUDIO_AFE>, + <&audiosys CLK_AUDIO_DAC>, + <&audiosys CLK_AUDIO_DAC_PREDIS>, + <&audiosys CLK_AUDIO_ADC>, + <&audiosys CLK_AUDIO_PDN_ADDA6_ADC>, + <&audiosys CLK_AUDIO_22M>, + <&audiosys CLK_AUDIO_24M>, + <&audiosys CLK_AUDIO_APLL_TUNER>, + <&audiosys CLK_AUDIO_APLL2_TUNER>, + <&audiosys CLK_AUDIO_I2S1>, + <&audiosys CLK_AUDIO_I2S2>, + <&audiosys CLK_AUDIO_I2S3>, + <&audiosys CLK_AUDIO_I2S4>, + <&audiosys CLK_AUDIO_TDM>, + <&audiosys CLK_AUDIO_TML>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_BCLK>, + <&topckgen CLK_TOP_MUX_AUDIO>, + <&topckgen CLK_TOP_MUX_AUD_INTBUS>, + <&topckgen CLK_TOP_SYSPLL_D2_D4>, + <&topckgen CLK_TOP_MUX_AUD_1>, + <&topckgen CLK_TOP_APLL1_CK>, + <&topckgen CLK_TOP_MUX_AUD_2>, + <&topckgen CLK_TOP_APLL2_CK>, + <&topckgen CLK_TOP_MUX_AUD_ENG1>, + <&topckgen CLK_TOP_APLL1_D8>, + <&topckgen CLK_TOP_MUX_AUD_ENG2>, + <&topckgen CLK_TOP_APLL2_D8>, + <&topckgen CLK_TOP_MUX_APLL_I2S0>, + <&topckgen CLK_TOP_MUX_APLL_I2S1>, + <&topckgen CLK_TOP_MUX_APLL_I2S2>, + <&topckgen CLK_TOP_MUX_APLL_I2S3>, + <&topckgen CLK_TOP_MUX_APLL_I2S4>, + <&topckgen CLK_TOP_MUX_APLL_I2S5>, + <&topckgen CLK_TOP_APLL12_DIV0>, + <&topckgen CLK_TOP_APLL12_DIV1>, + <&topckgen CLK_TOP_APLL12_DIV2>, + <&topckgen CLK_TOP_APLL12_DIV3>, + <&topckgen CLK_TOP_APLL12_DIV4>, + <&topckgen CLK_TOP_APLL12_DIVB>, + <&clk26m>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_adc_clk", + "aud_adc_adda6_clk", + "aud_apll22m_clk", + "aud_apll24m_clk", + "aud_apll1_tuner_clk", + "aud_apll2_tuner_clk", + "aud_i2s1_bclk_sw", + "aud_i2s2_bclk_sw", + "aud_i2s3_bclk_sw", + "aud_i2s4_bclk_sw", + "aud_tdm_clk", + "aud_tml_clk", + "aud_infra_clk", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_syspll_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_i2s3_m_sel", + "top_i2s4_m_sel", + "top_i2s5_m_sel", + "top_apll12_div0", + "top_apll12_div1", + "top_apll12_div2", + "top_apll12_div3", + "top_apll12_div4", + "top_apll12_divb", + "top_clk26m_clk"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8183_da7219.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8183_da7219.yaml new file mode 100644 index 000000000..b526e8123 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8183_da7219.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8183_da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8183 sound card with external codecs + +maintainers: + - Julien Massot <jmassot@collabora.com> + +description: + MediaTek MT8183 SoC-based sound cards with DA7219 as headset codec, + and MAX98357A, RT1015 or RT1015P as speaker amplifiers. Optionally includes HDMI codec. + +properties: + compatible: + enum: + - mediatek,mt8183_da7219_max98357 + - mediatek,mt8183_da7219_rt1015 + - mediatek,mt8183_da7219_rt1015p + + mediatek,headset-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the DA7219 headset codec. + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the MT8183 ASoC platform (e.g., AFE node). + + mediatek,hdmi-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Optional phandle to the HDMI codec (e.g., IT6505). + +required: + - compatible + - mediatek,headset-codec + - mediatek,platform + +additionalProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8183_da7219_max98357"; + mediatek,headset-codec = <&da7219>; + mediatek,hdmi-codec = <&it6505dptx>; + mediatek,platform = <&afe>; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8183_mt6358_ts3a227.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8183_mt6358_ts3a227.yaml new file mode 100644 index 000000000..43a6f9d40 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8183_mt6358_ts3a227.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8183_mt6358_ts3a227.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8183 sound card with MT6358, TS3A227, and MAX98357/RT1015 codecs + +maintainers: + - Julien Massot <julien.massot@collabora.com> + +description: + MediaTek MT8183 SoC-based sound cards using the MT6358 codec, + with optional TS3A227 headset codec, EC codec (via Chrome EC), and HDMI audio. + Speaker amplifier can be one of MAX98357A/B, RT1015, or RT1015P. + +properties: + compatible: + enum: + - mediatek,mt8183_mt6358_ts3a227_max98357 + - mediatek,mt8183_mt6358_ts3a227_max98357b + - mediatek,mt8183_mt6358_ts3a227_rt1015 + - mediatek,mt8183_mt6358_ts3a227_rt1015p + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the MT8183 ASoC platform node (e.g., AFE). + + mediatek,headset-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the TS3A227 headset codec. + + mediatek,ec-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: | + Optional phandle to a ChromeOS EC codec node. + See bindings in google,cros-ec-codec.yaml. + + mediatek,hdmi-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Optional phandle to an HDMI audio codec node. + +required: + - compatible + - mediatek,platform + +additionalProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8183_mt6358_ts3a227_max98357"; + mediatek,headset-codec = <&ts3a227>; + mediatek,ec-codec = <&ec_codec>; + mediatek,hdmi-codec = <&it6505dptx>; + mediatek,platform = <&afe>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml new file mode 100644 index 000000000..77af276ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml @@ -0,0 +1,241 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8188-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AFE PCM controller for mt8188 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8188-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + memory-region: + maxItems: 1 + description: | + Shared memory region for AFE memif. A "shared-dma-pool". + See dtschema reserved-memory/shared-dma-pool.yaml for details. + + mediatek,topckgen: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the mediatek topckgen controller + + mediatek,infracfg: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the mediatek infracfg controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: 26M clock + - description: audio pll1 clock + - description: audio pll2 clock + - description: clock divider for i2si1_mck + - description: clock divider for i2si2_mck + - description: clock divider for i2so1_mck + - description: clock divider for i2so2_mck + - description: clock divider for dptx_mck + - description: a1sys hoping clock + - description: audio intbus clock + - description: audio hires clock + - description: audio local bus clock + - description: mux for dptx_mck + - description: mux for i2so1_mck + - description: mux for i2so2_mck + - description: mux for i2si1_mck + - description: mux for i2si2_mck + - description: audio 26m clock + - description: audio pll1 divide 4 + - description: audio pll2 divide 4 + - description: clock divider for iec + - description: mux for a2sys clock + - description: mux for aud_iec + + clock-names: + items: + - const: clk26m + - const: apll1 + - const: apll2 + - const: apll12_div0 + - const: apll12_div1 + - const: apll12_div2 + - const: apll12_div3 + - const: apll12_div9 + - const: top_a1sys_hp + - const: top_aud_intbus + - const: top_audio_h + - const: top_audio_local_bus + - const: top_dptx + - const: top_i2so1 + - const: top_i2so2 + - const: top_i2si1 + - const: top_i2si2 + - const: adsp_audio_26m + - const: apll1_d4 + - const: apll2_d4 + - const: apll12_div4 + - const: top_a2sys + - const: top_aud_iec + + mediatek,etdm-in1-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in module. + enum: + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + mediatek,etdm-in2-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in module. + enum: + - 0 # etdm1_in + - 2 # etdm1_out + - 3 # etdm2_out + + mediatek,etdm-out1-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 3 # etdm2_out + + mediatek,etdm-out2-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + +patternProperties: + "^mediatek,etdm-in[1-2]-chn-disabled$": + $ref: /schemas/types.yaml#/definitions/uint8-array + minItems: 1 + maxItems: 16 + description: + This is a list of channel IDs which should be disabled. + By default, all data received from ETDM pins will be outputted to + memory. etdm in supports disable_out in direct mode(w/o interconn), + so user can disable the specified channels by the property. + uniqueItems: true + items: + minimum: 0 + maximum: 15 + + "^mediatek,etdm-in[1-2]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-out[1-3]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + +required: + - compatible + - reg + - interrupts + - resets + - reset-names + - mediatek,topckgen + - mediatek,infracfg + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe@10b10000 { + compatible = "mediatek,mt8188-afe"; + reg = <0x10b10000 0x10000>; + interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; + resets = <&watchdog 14>; + reset-names = "audiosys"; + memory-region = <&snd_dma_mem_reserved>; + mediatek,topckgen = <&topckgen>; + mediatek,infracfg = <&infracfg_ao>; + power-domains = <&spm 13>; //MT8188_POWER_DOMAIN_AUDIO + mediatek,etdm-in2-cowork-source = <2>; + mediatek,etdm-out2-cowork-source = <0>; + mediatek,etdm-in1-multi-pin-mode; + mediatek,etdm-in1-chn-disabled = /bits/ 8 <0x0 0x2>; + clocks = <&clk26m>, + <&apmixedsys 9>, //CLK_APMIXED_APLL1 + <&apmixedsys 10>, //CLK_APMIXED_APLL2 + <&topckgen 186>, //CLK_TOP_APLL12_CK_DIV0 + <&topckgen 187>, //CLK_TOP_APLL12_CK_DIV1 + <&topckgen 188>, //CLK_TOP_APLL12_CK_DIV2 + <&topckgen 189>, //CLK_TOP_APLL12_CK_DIV3 + <&topckgen 191>, //CLK_TOP_APLL12_CK_DIV9 + <&topckgen 83>, //CLK_TOP_A1SYS_HP + <&topckgen 31>, //CLK_TOP_AUD_INTBUS + <&topckgen 32>, //CLK_TOP_AUDIO_H + <&topckgen 69>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 81>, //CLK_TOP_DPTX + <&topckgen 77>, //CLK_TOP_I2SO1 + <&topckgen 78>, //CLK_TOP_I2SO2 + <&topckgen 79>, //CLK_TOP_I2SI1 + <&topckgen 80>, //CLK_TOP_I2SI2 + <&adsp_audio26m 0>, //CLK_AUDIODSP_AUDIO26M + <&topckgen 132>, //CLK_TOP_APLL1_D4 + <&topckgen 133>, //CLK_TOP_APLL2_D4 + <&topckgen 183>, //CLK_TOP_APLL12_CK_DIV4 + <&topckgen 84>, //CLK_TOP_A2SYS + <&topckgen 82>; //CLK_TOP_AUD_IEC>; + clock-names = "clk26m", + "apll1", + "apll2", + "apll12_div0", + "apll12_div1", + "apll12_div2", + "apll12_div3", + "apll12_div9", + "top_a1sys_hp", + "top_aud_intbus", + "top_audio_h", + "top_audio_local_bus", + "top_dptx", + "top_i2so1", + "top_i2so2", + "top_i2si1", + "top_i2si2", + "adsp_audio_26m", + "apll1_d4", + "apll2_d4", + "apll12_div4", + "top_a2sys", + "top_aud_iec"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml new file mode 100644 index 000000000..539de75eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml @@ -0,0 +1,136 @@ +# 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: + oneOf: + - enum: + - mediatek,mt8188-es8326 + - mediatek,mt8188-mt6359-evb + - mediatek,mt8188-nau8825 + - mediatek,mt8188-rt5682s + - items: + - const: mediatek,mt8390-mt6359-evk + - const: mediatek,mt8188-mt6359-evb + + 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. + + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the MT8188 ADSP platform, which is the optional Audio DSP + hardware that provides additional audio functionalities if present. + The AFE will link to ADSP when the phandle is provided. + + mediatek,accdet: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle to the MT6359 accessory detection block, which detects audio + jack insertion and removal. This property should only be present if the + accdet block is actually wired to the audio jack pins and to be used for + jack detection. + +patternProperties: + "^dai-link-[0-9]+$": + 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 + - DL_SRC_BE + - UL_SRC_BE + - DMIC_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: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: + - cpu + - codec + + additionalProperties: false + + required: + - link-name + +required: + - compatible + - mediatek,platform + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8188-mt6359-evb"; + model = "MT6359-EVB"; + mediatek,platform = <&afe>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + audio-routing = + "Headphone", "Headphone L", + "Headphone", "Headphone R", + "AIN1", "Headset Mic"; + dai-link-0 { + link-name = "ETDM3_OUT_BE"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&hdmi0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8189-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8189-afe-pcm.yaml new file mode 100644 index 000000000..9c9f21652 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8189-afe-pcm.yaml @@ -0,0 +1,178 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8189-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End PCM controller for MT8189 + +maintainers: + - Darren Ye <darren.ye@mediatek.com> + - Cyril Chao <cyril.chao@mediatek.com> + +properties: + compatible: + const: mediatek,mt8189-afe-pcm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + memory-region: + maxItems: 1 + + mediatek,apmixedsys: + $ref: /schemas/types.yaml#/definitions/phandle + description: To set up the apll12 tuner + + power-domains: + maxItems: 1 + + clocks: + items: + - description: mux for audio intbus + - description: mux for audio engen1 + - description: mux for audio engen2 + - description: mux for audio h + - description: audio apll1 clock + - description: audio apll2 clock + - description: audio apll1 divide4 + - description: audio apll2 divide4 + - description: audio apll12 divide for i2sin0 + - description: audio apll12 divide for i2sin1 + - description: audio apll12 divide for i2sout0 + - description: audio apll12 divide for i2sout1 + - description: audio apll12 divide for fmi2s + - description: audio apll12 divide for tdmout mck + - description: audio apll12 divide for tdmout bck + - description: mux for audio apll1 + - description: mux for audio apll2 + - description: mux for i2sin0 mck + - description: mux for i2sin1 mck + - description: mux for i2sout0 mck + - description: mux for i2sout1 mck + - description: mux for fmi2s mck + - description: mux for tdmout mck + - description: 26m clock + - description: audio slv clock + - description: audio mst clock + - description: audio intbus clock + + clock-names: + items: + - const: top_aud_intbus + - const: top_aud_eng1 + - const: top_aud_eng2 + - const: top_aud_h + - const: apll1 + - const: apll2 + - const: apll1_d4 + - const: apll2_d4 + - const: apll12_div_i2sin0 + - const: apll12_div_i2sin1 + - const: apll12_div_i2sout0 + - const: apll12_div_i2sout1 + - const: apll12_div_fmi2s + - const: apll12_div_tdmout_m + - const: apll12_div_tdmout_b + - const: top_apll1 + - const: top_apll2 + - const: top_i2sin0 + - const: top_i2sin1 + - const: top_i2sout0 + - const: top_i2sout1 + - const: top_fmi2s + - const: top_dptx + - const: clk26m + - const: aud_slv_ck_peri + - const: aud_mst_ck_peri + - const: aud_intbus_ck_peri + +required: + - compatible + - reg + - interrupts + - memory-region + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + soc { + #address-cells = <2>; + #size-cells = <2>; + + afe@11050000 { + compatible = "mediatek,mt8189-afe-pcm"; + reg = <0 0x11050000 0 0x10000>; + interrupts = <GIC_SPI 392 IRQ_TYPE_LEVEL_HIGH 0>; + memory-region = <&afe_dma_mem_reserved>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + power-domains = <&scpsys 1>; //MT8189_POWER_DOMAIN_AUDIO + clocks = <&topckgen_clk 23>, //CLK_TOP_AUD_INTBUS_SEL + <&topckgen_clk 39>, //CLK_TOP_AUD_ENGEN1_SEL + <&topckgen_clk 40>, //CLK_TOP_AUD_ENGEN2_SEL + <&topckgen_clk 49>, //CLK_TOP_AUDIO_H_SEL + <&topckgen_clk 146>, //CLK_TOP_APLL1 + <&topckgen_clk 151>, //CLK_TOP_APLL2 + <&topckgen_clk 148>, //CLK_TOP_APLL1_D4 + <&topckgen_clk 153>, //CLK_TOP_APLL2_D4 + <&topckgen_clk 93>, //CLK_TOP_APLL12_CK_DIV_I2SIN0 + <&topckgen_clk 94>, //CLK_TOP_APLL12_CK_DIV_I2SIN1 + <&topckgen_clk 95>, //CLK_TOP_APLL12_CK_DIV_I2SOUT0 + <&topckgen_clk 96>, //CLK_TOP_APLL12_CK_DIV_I2SOUT1 + <&topckgen_clk 97>, //CLK_TOP_APLL12_CK_DIV_FMI2S + <&topckgen_clk 98>, //CLK_TOP_APLL12_CK_DIV_TDMOUT_M + <&topckgen_clk 99>, //CLK_TOP_APLL12_CK_DIV_TDMOUT_B + <&topckgen_clk 44>, //CLK_TOP_AUD_1_SEL + <&topckgen_clk 45>, //CLK_TOP_AUD_2_SEL + <&topckgen_clk 78>, //CLK_TOP_APLL_I2SIN0_MCK_SEL + <&topckgen_clk 79>, //CLK_TOP_APLL_I2SIN1_MCK_SEL + <&topckgen_clk 84>, //CLK_TOP_APLL_I2SOUT0_MCK_SEL + <&topckgen_clk 85>, //CLK_TOP_APLL_I2SOUT1_MCK_SEL + <&topckgen_clk 90>, //CLK_TOP_APLL_FMI2S_MCK_SEL + <&topckgen_clk 91>, //CLK_TOP_APLL_TDMOUT_MCK_SEL + <&topckgen_clk 191>, //CLK_TOP_TCK_26M_MX9 + <&pericfg_ao_clk 77>, //CLK_PERAO_AUDIO0 + <&pericfg_ao_clk 78>, //CLK_PERAO_AUDIO1 + <&pericfg_ao_clk 79>; //CLK_PERAO_AUDIO2 + clock-names = "top_aud_intbus", + "top_aud_eng1", + "top_aud_eng2", + "top_aud_h", + "apll1", + "apll2", + "apll1_d4", + "apll2_d4", + "apll12_div_i2sin0", + "apll12_div_i2sin1", + "apll12_div_i2sout0", + "apll12_div_i2sout1", + "apll12_div_fmi2s", + "apll12_div_tdmout_m", + "apll12_div_tdmout_b", + "top_apll1", + "top_apll2", + "top_i2sin0", + "top_i2sin1", + "top_i2sout0", + "top_i2sout1", + "top_fmi2s", + "top_dptx", + "clk26m", + "aud_slv_ck_peri", + "aud_mst_ck_peri", + "aud_intbus_ck_peri"; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8189-nau8825.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8189-nau8825.yaml new file mode 100644 index 000000000..dd9ee0a3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8189-nau8825.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8189-nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8189 ASoC sound card + +maintainers: + - Darren Ye <darren.ye@mediatek.com> + - Cyril Chao <cyril.chao@mediatek.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8189-nau8825 + - mediatek,mt8189-rt5650 + - mediatek,mt8189-rt5682s + - mediatek,mt8189-rt5682i + - mediatek,mt8189-es8326 + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8189 ASoC platform. + +patternProperties: + "^dai-link-[0-9]+$": + 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. + enum: + - TDM_DPTX_BE + - I2SOUT0_BE + - I2SIN0_BE + - I2SOUT1_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. + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: + - cpu + - codec + + additionalProperties: false + + required: + - link-name + +required: + - compatible + - mediatek,platform + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8189-nau8825"; + model = "mt8189_rt9123_8825"; + mediatek,platform = <&afe>; + dai-link-0 { + link-name = "I2SOUT1_BE"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&nau8825>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8196-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8196-afe.yaml new file mode 100644 index 000000000..949f8622b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8196-afe.yaml @@ -0,0 +1,113 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8196-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End PCM controller for MT8196 + +maintainers: + - Darren Ye <darren.ye@mediatek.com> + +properties: + compatible: + const: mediatek,mt8196-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + memory-region: + maxItems: 1 + + power-domains: + maxItems: 1 + + clocks: + items: + - description: mux for audio intbus + - description: mux for audio engen1 + - description: mux for audio engen2 + - description: mux for audio h + - description: audio apll1 clock + - description: audio apll2 clock + - description: audio apll12 divide for i2sin0 + - description: audio apll12 divide for i2sin1 + - description: audio apll12 divide for fmi2s + - description: audio apll12 divide for tdmout mck + - description: audio apll12 divide for tdmout bck + - description: mux for adsp clock + + clock-names: + items: + - const: top_aud_intbus + - const: top_aud_eng1 + - const: top_aud_eng2 + - const: top_aud_h + - const: apll1 + - const: apll2 + - const: apll12_div_i2sin0 + - const: apll12_div_i2sin1 + - const: apll12_div_fmi2s + - const: apll12_div_tdmout_m + - const: apll12_div_tdmout_b + - const: top_adsp + +required: + - compatible + - reg + - interrupts + - memory-region + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + soc { + #address-cells = <2>; + #size-cells = <2>; + + afe@1a110000 { + compatible = "mediatek,mt8196-afe"; + reg = <0 0x1a110000 0 0x9000>; + interrupts = <GIC_SPI 351 IRQ_TYPE_LEVEL_HIGH 0>; + memory-region = <&afe_dma_mem_reserved>; + power-domains = <&scpsys 14>; //MT8196_POWER_DOMAIN_AUDIO + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + clocks = <&vlp_cksys_clk 40>, //CLK_VLP_CK_AUD_INTBUS_SEL + <&vlp_cksys_clk 38>, //CLK_VLP_CK_AUD_ENGEN1_SEL + <&vlp_cksys_clk 39>, //CLK_VLP_CK_AUD_ENGEN2_SEL + <&vlp_cksys_clk 37>, //CLK_VLP_CK_AUDIO_H_SEL + <&vlp_cksys_clk 0>, //CLK_VLP_CK_VLP_APLL1 + <&vlp_cksys_clk 1>, //CLK_VLP_CK_VLP_APLL2 + <&cksys_clk 80>, //CLK_CK_APLL12_CK_DIV_I2SIN0 + <&cksys_clk 81>, //CLK_CK_APLL12_CK_DIV_I2SIN1 + <&cksys_clk 92>, //CLK_CK_APLL12_CK_DIV_FMI2S + <&cksys_clk 93>, //CLK_CK_APLL12_CK_DIV_TDMOUT_M + <&cksys_clk 94>, //CLK_CK_APLL12_CK_DIV_TDMOUT_B + <&cksys_clk 45>; //CLK_CK_ADSP_SEL + clock-names = "top_aud_intbus", + "top_aud_eng1", + "top_aud_eng2", + "top_aud_h", + "apll1", + "apll2", + "apll12_div_i2sin0", + "apll12_div_i2sin1", + "apll12_div_fmi2s", + "apll12_div_tdmout_m", + "apll12_div_tdmout_b", + "top_adsp"; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8196-nau8825.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8196-nau8825.yaml new file mode 100644 index 000000000..83350faa1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8196-nau8825.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/mediatek,mt8196-nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8196 ASoC sound card + +maintainers: + - Darren Ye <darren.ye@mediatek.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8196-nau8825-sound + - mediatek,mt8196-rt5682s-sound + - mediatek,mt8196-rt5650-sound + + 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: + - TDM_DPTX_BE + - I2SOUT6_BE + - I2SIN6_BE + - I2SOUT4_BE + - I2SOUT3_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: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: + - cpu + - codec + + additionalProperties: false + + required: + - link-name + +required: + - compatible + - mediatek,platform + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8196-nau8825-sound"; + model = "mt8196-nau8825"; + mediatek,platform = <&afe>; + dai-link-0 { + link-name = "I2SOUT6_BE"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&nau8825>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml new file mode 100644 index 000000000..45ad56d37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml @@ -0,0 +1,130 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8365-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End PCM controller for MT8365 + +maintainers: + - Alexandre Mergnat <amergnat@baylibre.com> + +properties: + compatible: + const: mediatek,mt8365-afe-pcm + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: 26M clock + - description: mux for audio clock + - description: audio i2s0 mck + - description: audio i2s1 mck + - description: audio i2s2 mck + - description: audio i2s3 mck + - description: engen 1 clock + - description: engen 2 clock + - description: audio 1 clock + - description: audio 2 clock + - description: mux for i2s0 + - description: mux for i2s1 + - description: mux for i2s2 + - description: mux for i2s3 + + clock-names: + items: + - const: top_clk26m_clk + - const: top_audio_sel + - const: audio_i2s0_m + - const: audio_i2s1_m + - const: audio_i2s2_m + - const: audio_i2s3_m + - const: engen1 + - const: engen2 + - const: aud1 + - const: aud2 + - const: i2s0_m_sel + - const: i2s1_m_sel + - const: i2s2_m_sel + - const: i2s3_m_sel + + interrupts: + maxItems: 1 + + power-domains: + maxItems: 1 + + mediatek,dmic-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Indicates how many data pins are used to transmit two channels of PDM + signal. 1 means two wires, 0 means one wire. Default value is 0. + enum: + - 0 # one wire + - 1 # two wires + +required: + - compatible + - reg + - clocks + - clock-names + - interrupts + - power-domains + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mediatek,mt8365-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mediatek,mt8365-power.h> + + soc { + #address-cells = <2>; + #size-cells = <2>; + + audio-controller@11220000 { + compatible = "mediatek,mt8365-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + #sound-dai-cells = <0>; + clocks = <&clk26m>, + <&topckgen CLK_TOP_AUDIO_SEL>, + <&topckgen CLK_TOP_AUD_I2S0_M>, + <&topckgen CLK_TOP_AUD_I2S1_M>, + <&topckgen CLK_TOP_AUD_I2S2_M>, + <&topckgen CLK_TOP_AUD_I2S3_M>, + <&topckgen CLK_TOP_AUD_ENGEN1_SEL>, + <&topckgen CLK_TOP_AUD_ENGEN2_SEL>, + <&topckgen CLK_TOP_AUD_1_SEL>, + <&topckgen CLK_TOP_AUD_2_SEL>, + <&topckgen CLK_TOP_APLL_I2S0_SEL>, + <&topckgen CLK_TOP_APLL_I2S1_SEL>, + <&topckgen CLK_TOP_APLL_I2S2_SEL>, + <&topckgen CLK_TOP_APLL_I2S3_SEL>; + clock-names = "top_clk26m_clk", + "top_audio_sel", + "audio_i2s0_m", + "audio_i2s1_m", + "audio_i2s2_m", + "audio_i2s3_m", + "engen1", + "engen2", + "aud1", + "aud2", + "i2s0_m_sel", + "i2s1_m_sel", + "i2s2_m_sel", + "i2s3_m_sel"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_LOW>; + power-domains = <&spm MT8365_POWER_DOMAIN_AUDIO>; + mediatek,dmic-mode = <1>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml new file mode 100644 index 000000000..ff9ebb63a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml @@ -0,0 +1,107 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8365-mt6357.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8365 ASoC sound card + +maintainers: + - Alexandre Mergnat <amergnat@baylibre.com> + +properties: + compatible: + const: mediatek,mt8365-mt6357 + + pinctrl-names: + minItems: 1 + items: + - const: default + - const: dmic + - const: miso_off + - const: miso_on + - const: mosi_off + - const: mosi_on + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8365 ASoC platform. + +patternProperties: + "^dai-link-[0-9]+$": + type: object + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + codec: + type: object + description: Holds subnode which indicates codec dai. + + properties: + sound-dai: + maxItems: 1 + description: phandle of the codec DAI + + additionalProperties: false + + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - I2S_IN_BE + - I2S_OUT_BE + - PCM1_BE + - PDM1_BE + - PDM2_BE + - PDM3_BE + - PDM4_BE + - SPDIF_IN_BE + - SPDIF_OUT_BE + - TDM_IN_BE + - TDM_OUT_BE + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + required: + - link-name + - sound-dai + + additionalProperties: false + +required: + - compatible + - pinctrl-names + - mediatek,platform + +additionalProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8365-mt6357"; + pinctrl-names = "default", + "dmic", + "miso_off", + "miso_on", + "mosi_off", + "mosi_on"; + pinctrl-0 = <&aud_default_pins>; + pinctrl-1 = <&aud_dmic_pins>; + pinctrl-2 = <&aud_miso_off_pins>; + pinctrl-3 = <&aud_miso_on_pins>; + pinctrl-4 = <&aud_mosi_off_pins>; + pinctrl-5 = <&aud_mosi_on_pins>; + mediatek,platform = <&afe>; + + /* hdmi interface */ + dai-link-0 { + link-name = "I2S_OUT_BE"; + sound-dai = <&afe>; + + codec { + sound-dai = <&it66121hdmitx>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mtk-btcvsd-snd.yaml b/Documentation/devicetree/bindings/sound/mediatek,mtk-btcvsd-snd.yaml new file mode 100644 index 000000000..1b7451655 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mtk-btcvsd-snd.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mtk-btcvsd-snd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek ALSA BT SCO CVSD/MSBC + +maintainers: + - Luca Leonardo Scorcia <l.scorcia@gmail.com> + +properties: + compatible: + const: mediatek,mtk-btcvsd-snd + + reg: + items: + - description: PKV region + - description: SRAM_BANK2 region + + interrupts: + items: + - description: BT-SCO interrupt + + mediatek,infracfg: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the infracfg controller + + mediatek,offset: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Array of register offsets and masks + items: + - description: infra_misc_offset + - description: infra_conn_bt_cvsd_mask + - description: cvsd_mcu_read_offset + - description: cvsd_mcu_write_offset + - description: cvsd_packet_indicator_offset + +required: + - compatible + - reg + - interrupts + - mediatek,infracfg + - mediatek,offset + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + + mtk-btcvsd-snd@18000000 { + compatible = "mediatek,mtk-btcvsd-snd"; + reg = <0x18000000 0x1000>, + <0x18080000 0x8000>; + interrupts = <GIC_SPI 286 IRQ_TYPE_LEVEL_LOW>; + mediatek,infracfg = <&infrasys>; + mediatek,offset = <0xf00 0x800 0xfd0 0xfd4 0xfd8>; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml new file mode 100644 index 000000000..fb630a184 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-i2smcc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip I2S Multi-Channel Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and + supports a Time Division Multiplexed (TDM) interface with external + multi-channel audio codecs. It consists of a receiver, a transmitter and a + common clock generator that can be enabled separately to provide Adapter, + Client or Controller modes with receiver and/or transmitter active. + On later I2SMCC versions (starting with Microchip's SAMA7G5) I2S + multi-channel is supported by using multiple data pins, output and + input, without TDM. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - enum: + - microchip,sam9x60-i2smcc + - microchip,sama7g5-i2smcc + - items: + - enum: + - microchip,sam9x7-i2smcc + - const: microchip,sam9x60-i2smcc + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock (Optional). Should be set mostly when Master + Mode is required. + minItems: 1 + + clock-names: + items: + - const: pclk + - const: gclk + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + microchip,tdm-data-pair: + description: + Represents the DIN/DOUT pair pins that are used to receive/send + TDM data. It is optional and it is only needed if the controller + uses the TDM mode. + $ref: /schemas/types.yaml#/definitions/uint8 + enum: [0, 1, 2, 3] + default: 0 + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + const: microchip,sam9x60-i2smcc + then: + properties: + microchip,tdm-data-pair: false + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f001c000 { + #sound-dai-cells = <0>; + compatible = "microchip,sam9x60-i2smcc"; + reg = <0xf001c000 0x100>; + interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(36))>, + <&dma0 (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(37))>; + dma-names = "tx", "rx"; + clocks = <&i2s_clk>, <&i2s_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml new file mode 100644 index 000000000..9aa65c975 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-pdmc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip Pulse Density Microphone Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Pulse Density Microphone Controller (PDMC) interfaces up to 4 + digital microphones having Pulse Density Modulated (PDM) outputs. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: microchip,sama7g5-pdmc + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + + microchip,mic-pos: + description: | + Position of PDM microphones on the DS line and the sampling edge (rising + or falling) of the CLK line. A microphone is represented as a pair of DS + line and the sampling edge. The first microphone is mapped to channel 0, + the second to channel 1, etc. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: value for DS line + enum: [0, 1] + - description: value for sampling edge + enum: [0, 1] + minItems: 1 + maxItems: 4 + uniqueItems: true + + microchip,startup-delay-us: + description: | + Specifies the delay in microseconds that needs to be applied after + enabling the PDMC microphones to avoid unwanted noise due to microphones + not being ready. + +required: + - compatible + - reg + - "#sound-dai-cells" + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - microchip,mic-pos + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/sound/microchip,pdmc.h> + + pdmc: sound@e1608000 { + compatible = "microchip,sama7g5-pdmc"; + reg = <0xe1608000 0x4000>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 68 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(37)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 68>, <&pmc PMC_TYPE_GCK 68>; + clock-names = "pclk", "gclk"; + microchip,mic-pos = <MCHP_PDMC_DS0 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS0 MCHP_PDMC_CLK_NEGATIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_POSITIVE>, + <MCHP_PDMC_DS1 MCHP_PDMC_CLK_NEGATIVE>; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml new file mode 100644 index 000000000..7fbab5871 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + +allOf: + - $ref: dai-common.yaml# + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdifrx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: RX DMA Channel + maxItems: 1 + + dma-names: + const: rx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdifrx: spdifrx@e1614000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdifrx"; + reg = <0xe1614000 0x4000>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(49)>; + dma-names = "rx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 84>, <&pmc PMC_TYPE_GCK 84>; + clock-names = "pclk", "gclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml new file mode 100644 index 000000000..4702c5287 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml @@ -0,0 +1,78 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu@microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a serial port + compliant with the IEC-60958 standard. + +allOf: + - $ref: dai-common.yaml# + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + maxItems: 1 + + dma-names: + const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt new file mode 100644 index 000000000..912f8fae1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt @@ -0,0 +1,23 @@ +Mikroe-PROTO audio board + +Required properties: + - compatible: "mikroe,mikroe-proto" + - dai-format: Must be "i2s". + - i2s-controller: The phandle of the I2S controller. + - audio-codec: The phandle of the WM8731 audio codec. +Optional properties: + - model: The user-visible name of this sound complex. + - bitclock-master: Indicates dai-link bit clock master; for details see simple-card.txt (1). + - frame-master: Indicates dai-link frame master; for details see simple-card.txt (1). + +(1) : There must be the same master for both bit and frame clocks. + +Example: + sound { + compatible = "mikroe,mikroe-proto"; + model = "wm8731 @ sama5d2_xplained"; + i2s-controller = <&i2s0>; + audio-codec = <&wm8731>; + dai-format = "i2s"; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt new file mode 100644 index 000000000..feef39b4a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -0,0 +1,34 @@ +Marvell PXA SSP CPU DAI bindings + +Required properties: + + compatible Must be "mrvl,pxa-ssp-dai" + port A phandle reference to a PXA ssp upstream device + +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + +Example: + + /* upstream device */ + + ssp1: ssp@41000000 { + compatible = "mrvl,pxa3xx-ssp"; + reg = <0x41000000 0x40>; + interrupts = <24>; + clock-names = "pxa27x-ssp.0"; + }; + + /* DAI as user */ + + ssp_dai0: ssp_dai@0 { + compatible = "mrvl,pxa-ssp-dai"; + port = <&ssp1>; + #sound-dai-cells = <0>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mscc,zl38060.yaml b/Documentation/devicetree/bindings/sound/mscc,zl38060.yaml new file mode 100644 index 000000000..994313fd1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mscc,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/mscc,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>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt new file mode 100644 index 000000000..05574446c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt @@ -0,0 +1,43 @@ +MT2701 with CS42448 CODEC + +Required properties: +- compatible: "mediatek,mt2701-cs42448-machine" +- mediatek,platform: the phandle of MT2701 ASoC platform +- audio-routing: a list of the connections between audio +- mediatek,audio-codec: the phandles of cs42448 codec +- mediatek,audio-codec-bt-mrg the phandles of bt-sco dummy codec +- pinctrl-names: Should contain only one value - "default" +- pinctrl-0: Should specify pin control groups used for this controller. +- i2s1-in-sel-gpio1, i2s1-in-sel-gpio2: Should specify two gpio pins to + control I2S1-in mux. + +Example: + + sound:sound { + compatible = "mediatek,mt2701-cs42448-machine"; + mediatek,platform = <&afe>; + /* CS42448 Machine name */ + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "AMIC", + "AIN1R", "AMIC", + "AIN2L", "Tuner In", + "AIN2R", "Tuner In", + "AIN3L", "Satellite Tuner In", + "AIN3R", "Satellite Tuner In", + "AIN3L", "AUX In", + "AIN3R", "AUX In"; + mediatek,audio-codec = <&cs42448>; + mediatek,audio-codec-bt-mrg = <&bt_sco_codec>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + i2s1-in-sel-gpio1 = <&pio 53 0>; + i2s1-in-sel-gpio2 = <&pio 54 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt6358.txt b/Documentation/devicetree/bindings/sound/mt6358.txt new file mode 100644 index 000000000..fbe9e55c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6358.txt @@ -0,0 +1,26 @@ +Mediatek MT6358 Audio Codec + +The communication between MT6358 and SoC is through Mediatek PMIC wrapper. +For more detail, please visit Mediatek PMIC wrapper documentation. + +Must be a child node of PMIC wrapper. + +Required properties: + +- compatible - "string" - One of: + "mediatek,mt6358-sound" + "mediatek,mt6366-sound" +- Avdd-supply : power source of AVDD + +Optional properties: +- mediatek,dmic-mode : Indicates how many data pins are used to transmit two + channels of PDM signal. 0 means two wires, 1 means one wire. Default + value is 0. + +Example: + +mt6358_snd { + compatible = "mediatek,mt6358-sound"; + Avdd-supply = <&mt6358_vaud28_reg>; + mediatek,dmic-mode = <0>; +}; diff --git a/Documentation/devicetree/bindings/sound/mt6359.yaml b/Documentation/devicetree/bindings/sound/mt6359.yaml new file mode 100644 index 000000000..128698630 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6359.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt6359.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek MT6359 Codec + +maintainers: + - Eason Yen <eason.yen@mediatek.com> + - Jiaxin Yu <jiaxin.yu@mediatek.com> + - Shane Chien <shane.chien@mediatek.com> + +description: | + The communication between MT6359 and SoC is through Mediatek PMIC wrapper. + For more detail, please visit Mediatek PMIC wrapper documentation. + Must be a child node of PMIC wrapper. + +properties: + mediatek,dmic-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Indicates how many data pins are used to transmit two channels of PDM + signal. 0 means two wires, 1 means one wire. Default value is 0. + enum: + - 0 # two wires + - 1 # one wire + + 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: audio-codec { + mediatek,dmic-mode = <0>; + mediatek,mic-type-0 = <2>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt new file mode 100644 index 000000000..0ae29de15 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt @@ -0,0 +1,42 @@ +Mediatek AFE PCM controller for mt6797 + +Required properties: +- compatible = "mediatek,mt6797-audio"; +- reg: register location and size +- interrupts: should contain AFE interrupt +- power-domains: should define the power domain +- clocks: Must contain an entry for each entry in clock-names +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "infra_sys_audio_26m", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_sys_pll1_d4", + "top_clk26m_clk"; + +Example: + + afe: mt6797-afe-pcm@11220000 { + compatible = "mediatek,mt6797-audio"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 151 IRQ_TYPE_LEVEL_LOW>; + power-domains = <&scpsys MT6797_POWER_DOMAIN_AUDIO>; + clocks = <&infrasys CLK_INFRA_AUDIO>, + <&infrasys CLK_INFRA_AUDIO_26M>, + <&infrasys CLK_INFRA_AUDIO_26M_PAD_TOP>, + <&topckgen CLK_TOP_MUX_AUDIO>, + <&topckgen CLK_TOP_MUX_AUD_INTBUS>, + <&topckgen CLK_TOP_SYSPLL3_D4>, + <&topckgen CLK_TOP_SYSPLL1_D4>, + <&clk26m>; + clock-names = "infra_sys_audio_clk", + "infra_sys_audio_26m", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_sys_pll1_d4", + "top_clk26m_clk"; + }; diff --git a/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt b/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt new file mode 100644 index 000000000..1d95a8840 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt @@ -0,0 +1,14 @@ +MT6797 with MT6351 CODEC + +Required properties: +- compatible: "mediatek,mt6797-mt6351-sound" +- mediatek,platform: the phandle of MT6797 ASoC platform +- mediatek,audio-codec: the phandles of MT6351 codec + +Example: + + sound { + compatible = "mediatek,mt6797-mt6351-sound"; + mediatek,audio-codec = <&mt6351_snd>; + mediatek,platform = <&afe>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt new file mode 100644 index 000000000..519e97c8f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -0,0 +1,15 @@ +MT8173 with MAX98090 CODEC + +Required properties: +- compatible : "mediatek,mt8173-max98090" +- mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-max98090"; + mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt new file mode 100644 index 000000000..ac28cdb49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -0,0 +1,16 @@ +MT8173 with RT5650 RT5676 CODECS and HDMI via I2S + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5676" +- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs + and of the hdmi encoder node +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5676"; + mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>; + mediatek,platform = <&afe>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt new file mode 100644 index 000000000..29dce2ac8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -0,0 +1,31 @@ +MT8173 with RT5650 CODECS and HDMI via I2S + +Required properties: +- compatible : "mediatek,mt8173-rt5650" +- mediatek,audio-codec: the phandles of rt5650 codecs + and of the hdmi encoder node +- mediatek,platform: the phandle of MT8173 ASoC platform + +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + +- mediatek,mclk: the MCLK source + 0 : external oscillator, MCLK = 12.288M + 1 : internal source from mt8173, MCLK = sampling rate*256 + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650"; + mediatek,audio-codec = <&rt5650 &hdmi0>; + mediatek,platform = <&afe>; + mediatek,mclk = <0>; + codec-capture { + sound-dai = <&rt5650 1>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml new file mode 100644 index 000000000..f5af2cf18 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml @@ -0,0 +1,180 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + memory-region: + description: memory region for audio DMA buffers + maxItems: 1 + + 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"; + memory-region = <&afe_dma_mem>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml new file mode 100644 index 000000000..037f21443 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml @@ -0,0 +1,197 @@ +# 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. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8186-mt6366-da7219-max98357-sound + + 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 the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - HDMI1 + - Headphones + - Line Out + - MIC + - Speakers + + # Sources + - Headset Mic + - HPL + - HPR + - Speaker + - TX + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8186 ASoC platform. + + headset-codec: + type: object + deprecated: true + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + playback-codecs: + type: object + deprecated: true + 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: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + items: + enum: + - I2S0 + - I2S1 + - I2S2 + - I2S3 + + 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: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: + - cpu + - codec + + required: + - link-name + +unevaluatedProperties: false + +required: + - compatible + - mediatek,platform + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false + +examples: + - | + + sound: mt8186-sound { + compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound"; + model = "mt8186_da7219_m98357"; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + mediatek,platform = <&afe>; + + audio-routing = + "Headphones", "HPL", + "Headphones", "HPR", + "MIC", "Headset Mic", + "Speakers", "Speaker", + "HDMI1", "TX"; + + hs-playback-dai-link { + link-name = "I2S0"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&da7219>; + }; + }; + + hs-capture-dai-link { + link-name = "I2S1"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&da7219>; + }; + }; + + spk-dp-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&anx_bridge_dp>, <&max98357a>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml new file mode 100644 index 000000000..ed93f18ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8186-mt6366-rt1019-rt5682s.yaml# +$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. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8186-mt6366-rt1019-rt5682s-sound + - mediatek,mt8186-mt6366-rt5682s-max98360-sound + - mediatek,mt8186-mt6366-rt5650-sound + + 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 the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - HDMI1 + - Headphone + - IN1P + - IN1N + - Line Out + - Speakers + + # Sources + - Headset Mic + - HPOL + - HPOR + - Speaker + - SPOL + - SPOR + - TX + + 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 + deprecated: true + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + playback-codecs: + type: object + deprecated: true + 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: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: [ I2S0, I2S1, I2S2, I2S3 ] + + 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 + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +unevaluatedProperties: false + +required: + - compatible + - mediatek,platform + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound: mt8186-sound { + compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound"; + model = "mt8186_rt1019_rt5682s"; + 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>; + mediatek,platform = <&afe>; + + dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>; + + audio-routing = + "Headphone", "HPOL", + "Headphone", "HPOR", + "IN1P", "Headset Mic", + "Speakers", "Speaker", + "HDMI1", "TX"; + + hs-playback-dai-link { + link-name = "I2S0"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682s 0>; + }; + }; + + hs-capture-dai-link { + link-name = "I2S1"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682s 0>; + }; + }; + + spk-hdmi-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&it6505dptx>, <&rt1019p>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000..16ae3328f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,253 @@ +# 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 + + memory-region: + description: memory region for audio DMA buffers + maxItems: 1 + + 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: ADDA ADC clock + - description: ADDA6 ADC clock + - description: Audio low-jitter 22.5792m clock + - description: Audio low-jitter 24.576m clock + - description: Audio PLL1 tuner clock + - description: Audio PLL2 tuner clock + - description: Audio Time-Division Multiplexing interface clock + - description: ADDA ADC Sine Generator clock + - description: audio Non-LE clock + - description: Audio DAC High-Resolution clock + - description: Audio High-Resolution ADC clock + - description: Audio High-Resolution ADC SineGen clock + - description: Audio ADDA6 High-Resolution ADC clock + - description: Tertiary ADDA DAC clock + - description: Tertiary ADDA DAC pre-distortion clock + - description: Tertiary ADDA DAC Sine Generator clock + - description: Tertiary ADDA DAC High-Resolution clock + - description: Audio infra sys clock + - description: Audio infra 26M clock + - description: Mux for audio clock + - description: Mux for audio internal bus clock + - description: Mux main divider by 4 + - description: Primary audio mux + - description: Primary audio PLL + - description: Secondary audio mux + - description: Secondary audio PLL + - description: Primary audio en-generator clock + - description: Primary PLL divider by 4 for IEC + - description: Secondary audio en-generator clock + - description: Secondary PLL divider by 4 for IEC + - description: Mux selector for I2S port 0 + - description: Mux selector for I2S port 1 + - description: Mux selector for I2S port 2 + - description: Mux selector for I2S port 3 + - description: Mux selector for I2S port 4 + - description: Mux selector for I2S port 5 + - description: Mux selector for I2S port 6 + - description: Mux selector for I2S port 7 + - description: Mux selector for I2S port 8 + - description: Mux selector for I2S port 9 + - description: APLL1 and APLL2 divider for I2S port 0 + - description: APLL1 and APLL2 divider for I2S port 1 + - description: APLL1 and APLL2 divider for I2S port 2 + - description: APLL1 and APLL2 divider for I2S port 3 + - description: APLL1 and APLL2 divider for I2S port 4 + - description: APLL1 and APLL2 divider for IEC + - description: APLL1 and APLL2 divider for I2S port 5 + - description: APLL1 and APLL2 divider for I2S port 6 + - description: APLL1 and APLL2 divider for I2S port 7 + - description: APLL1 and APLL2 divider for I2S port 8 + - description: APLL1 and APLL2 divider for I2S port 9 + - description: Top mux for audio subsystem + - description: 26MHz clock for audio subsystem + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_adc_clk + - const: aud_adda6_adc_clk + - const: aud_apll22m_clk + - const: aud_apll24m_clk + - const: aud_apll1_tuner_clk + - const: aud_apll2_tuner_clk + - const: aud_tdm_clk + - const: aud_tml_clk + - const: aud_nle + - const: aud_dac_hires_clk + - const: aud_adc_hires_clk + - const: aud_adc_hires_tml + - const: aud_adda6_adc_hires_clk + - const: aud_3rd_dac_clk + - const: aud_3rd_dac_predis_clk + - const: aud_3rd_dac_tml + - const: aud_3rd_dac_hires_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + - const: top_mux_audio + - const: top_mux_audio_int + - const: top_mainpll_d4_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_d4 + - const: top_mux_aud_eng2 + - const: top_apll2_d4 + - const: top_i2s0_m_sel + - const: top_i2s1_m_sel + - const: top_i2s2_m_sel + - const: top_i2s3_m_sel + - const: top_i2s4_m_sel + - const: top_i2s5_m_sel + - const: top_i2s6_m_sel + - const: top_i2s7_m_sel + - const: top_i2s8_m_sel + - const: top_i2s9_m_sel + - const: top_apll12_div0 + - const: top_apll12_div1 + - const: top_apll12_div2 + - const: top_apll12_div3 + - const: top_apll12_div4 + - const: top_apll12_divb + - const: top_apll12_div5 + - const: top_apll12_div6 + - const: top_apll12_div7 + - const: top_apll12_div8 + - const: top_apll12_div9 + - const: top_mux_audio_h + - const: top_clk26m_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>; + clocks = <&audsys CLK_AUD_AFE>, <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, <&audsys CLK_AUD_ADC>, + <&audsys CLK_AUD_ADDA6_ADC>, <&audsys CLK_AUD_22M>, + <&audsys CLK_AUD_24M>, <&audsys CLK_AUD_APLL_TUNER>, + <&audsys CLK_AUD_APLL2_TUNER>, <&audsys CLK_AUD_TDM>, + <&audsys CLK_AUD_TML>, <&audsys CLK_AUD_NLE>, + <&audsys CLK_AUD_DAC_HIRES>, <&audsys CLK_AUD_ADC_HIRES>, + <&audsys CLK_AUD_ADC_HIRES_TML>, <&audsys CLK_AUD_ADDA6_ADC_HIRES>, + <&audsys CLK_AUD_3RD_DAC>, <&audsys CLK_AUD_3RD_DAC_PREDIS>, + <&audsys CLK_AUD_3RD_DAC_TML>, <&audsys CLK_AUD_3RD_DAC_HIRES>, + <&infracfg CLK_INFRA_AUDIO>, <&infracfg CLK_INFRA_AUDIO_26M_B>, + <&topckgen CLK_TOP_AUDIO_SEL>, <&topckgen CLK_TOP_AUD_INTBUS_SEL>, + <&topckgen CLK_TOP_MAINPLL_D4_D4>, <&topckgen CLK_TOP_AUD_1_SEL>, + <&topckgen CLK_TOP_APLL1>, <&topckgen CLK_TOP_AUD_2_SEL>, + <&topckgen CLK_TOP_APLL2>, <&topckgen CLK_TOP_AUD_ENGEN1_SEL>, + <&topckgen CLK_TOP_APLL1_D4>, <&topckgen CLK_TOP_AUD_ENGEN2_SEL>, + <&topckgen CLK_TOP_APLL2_D4>, <&topckgen CLK_TOP_APLL_I2S0_M_SEL>, + <&topckgen CLK_TOP_APLL_I2S1_M_SEL>, <&topckgen CLK_TOP_APLL_I2S2_M_SEL>, + <&topckgen CLK_TOP_APLL_I2S3_M_SEL>, <&topckgen CLK_TOP_APLL_I2S4_M_SEL>, + <&topckgen CLK_TOP_APLL_I2S5_M_SEL>, <&topckgen CLK_TOP_APLL_I2S6_M_SEL>, + <&topckgen CLK_TOP_APLL_I2S7_M_SEL>, <&topckgen CLK_TOP_APLL_I2S8_M_SEL>, + <&topckgen CLK_TOP_APLL_I2S9_M_SEL>, <&topckgen CLK_TOP_APLL12_DIV0>, + <&topckgen CLK_TOP_APLL12_DIV1>, <&topckgen CLK_TOP_APLL12_DIV2>, + <&topckgen CLK_TOP_APLL12_DIV3>, <&topckgen CLK_TOP_APLL12_DIV4>, + <&topckgen CLK_TOP_APLL12_DIVB>, <&topckgen CLK_TOP_APLL12_DIV5>, + <&topckgen CLK_TOP_APLL12_DIV6>, <&topckgen CLK_TOP_APLL12_DIV7>, + <&topckgen CLK_TOP_APLL12_DIV8>, <&topckgen CLK_TOP_APLL12_DIV9>, + <&topckgen CLK_TOP_AUDIO_H_SEL>, <&clk26m>; + clock-names = "aud_afe_clk", "aud_dac_clk", + "aud_dac_predis_clk", "aud_adc_clk", + "aud_adda6_adc_clk", "aud_apll22m_clk", + "aud_apll24m_clk", "aud_apll1_tuner_clk", + "aud_apll2_tuner_clk", "aud_tdm_clk", + "aud_tml_clk", "aud_nle", + "aud_dac_hires_clk", "aud_adc_hires_clk", + "aud_adc_hires_tml", "aud_adda6_adc_hires_clk", + "aud_3rd_dac_clk", "aud_3rd_dac_predis_clk", + "aud_3rd_dac_tml", "aud_3rd_dac_hires_clk", + "aud_infra_clk", "aud_infra_26m_clk", + "top_mux_audio", "top_mux_audio_int", + "top_mainpll_d4_d4", "top_mux_aud_1", + "top_apll1_ck", "top_mux_aud_2", + "top_apll2_ck", "top_mux_aud_eng1", + "top_apll1_d4", "top_mux_aud_eng2", + "top_apll2_d4", "top_i2s0_m_sel", + "top_i2s1_m_sel", "top_i2s2_m_sel", + "top_i2s3_m_sel", "top_i2s4_m_sel", + "top_i2s5_m_sel", "top_i2s6_m_sel", + "top_i2s7_m_sel", "top_i2s8_m_sel", + "top_i2s9_m_sel", "top_apll12_div0", + "top_apll12_div1", "top_apll12_div2", + "top_apll12_div3", "top_apll12_div4", + "top_apll12_divb", "top_apll12_div5", + "top_apll12_div6", "top_apll12_div7", + "top_apll12_div8", "top_apll12_div9", + "top_mux_audio_h", "top_clk26m_clk"; + memory-region = <&afe_dma_mem>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml new file mode 100644 index 000000000..c4e68f31a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.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/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. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8192_mt6359_rt1015_rt5682 + - mediatek,mt8192_mt6359_rt1015p_rt5682 + - mediatek,mt8192_mt6359_rt1015p_rt5682s + + 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. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - Speakers + - Headphone Jack + - IN1P + - Left Spk + - Right Spk + + # Sources + - Headset Mic + - HPOL + - HPOR + - Left SPO + - Right SPO + - Speaker + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8192 ASoC platform. + + mediatek,hdmi-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of HDMI codec. + deprecated: true + + headset-codec: + type: object + additionalProperties: false + deprecated: true + + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + speaker-codecs: + type: object + additionalProperties: false + deprecated: true + + properties: + sound-dai: + minItems: 1 + maxItems: 2 + items: + maxItems: 1 + required: + - sound-dai + +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - I2S0 + - I2S1 + - I2S2 + - I2S3 + - I2S4 + - I2S5 + - I2S6 + - I2S7 + - I2S8 + - I2S9 + - TDM + + 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 + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +unevaluatedProperties: false + +required: + - compatible + - mediatek,platform + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false + mediatek,hdmi-codec: false + +examples: + - | + + sound: mt8192-sound { + compatible = "mediatek,mt8192_mt6359_rt1015_rt5682"; + model = "mt8192_mt6359_rt1015_rt5682"; + pinctrl-names = "aud_clk_mosi_off", + "aud_clk_mosi_on"; + pinctrl-0 = <&aud_clk_mosi_off>; + pinctrl-1 = <&aud_clk_mosi_on>; + mediatek,platform = <&afe>; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR", + "IN1P", "Headset Mic", + "Speakers", "Speaker"; + + spk-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt1015p>; + }; + }; + + hs-playback-dai-link { + link-name = "I2S8"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682 0>; + }; + }; + + hs-capture-dai-link { + link-name = "I2S9"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682 0>; + }; + }; + + displayport-dai-link { + link-name = "TDM"; + dai-format = "dsp_a"; + codec { + sound-dai = <&anx_bridge_dp>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml new file mode 100644 index 000000000..5c8dba2b3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml @@ -0,0 +1,200 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8195-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8195 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8195-audio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + memory-region: + maxItems: 1 + description: | + Shared memory region for AFE memif. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + + mediatek,topckgen: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: 26M clock + - description: audio pll1 clock + - description: audio pll2 clock + - description: clock divider for i2si1_mck + - description: clock divider for i2si2_mck + - description: clock divider for i2so1_mck + - description: clock divider for i2so2_mck + - description: clock divider for dptx_mck + - description: a1sys hoping clock + - description: audio intbus clock + - description: audio hires clock + - description: audio local bus clock + - description: mux for dptx_mck + - description: mux for i2so1_mck + - description: mux for i2so2_mck + - description: mux for i2si1_mck + - description: mux for i2si2_mck + - description: audio infra 26M clock + - description: infra bus clock + + clock-names: + items: + - const: clk26m + - const: apll1_ck + - const: apll2_ck + - const: apll12_div0 + - const: apll12_div1 + - const: apll12_div2 + - const: apll12_div3 + - const: apll12_div9 + - const: a1sys_hp_sel + - const: aud_intbus_sel + - const: audio_h_sel + - const: audio_local_bus_sel + - const: dptx_m_sel + - const: i2so1_m_sel + - const: i2so2_m_sel + - const: i2si1_m_sel + - const: i2si2_m_sel + - const: infra_ao_audio_26m_b + - const: scp_adsp_audiodsp + + mediatek,etdm-in1-chn-disabled: + $ref: /schemas/types.yaml#/definitions/uint8-array + maxItems: 24 + description: Specify which input channel should be disabled. + + mediatek,etdm-in2-chn-disabled: + $ref: /schemas/types.yaml#/definitions/uint8-array + maxItems: 16 + description: Specify which input channel should be disabled. + +patternProperties: + "^mediatek,etdm-in[1-2]-mclk-always-on-rate-hz$": + description: Specify etdm in mclk output rate for always on case. + + "^mediatek,etdm-out[1-3]-mclk-always-on-rate-hz$": + description: Specify etdm out mclk output rate for always on case. + + "^mediatek,etdm-in[1-2]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-out[1-3]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-in[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + "^mediatek,etdm-out[1-2]-cowork-source$": + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + +required: + - compatible + - reg + - interrupts + - resets + - reset-names + - mediatek,topckgen + - power-domains + - clocks + - clock-names + - memory-region + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe: mt8195-afe-pcm@10890000 { + compatible = "mediatek,mt8195-audio"; + reg = <0x10890000 0x10000>; + interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; + resets = <&watchdog 14>; + reset-names = "audiosys"; + mediatek,topckgen = <&topckgen>; + power-domains = <&spm 7>; //MT8195_POWER_DOMAIN_AUDIO + memory-region = <&snd_dma_mem_reserved>; + clocks = <&clk26m>, + <&topckgen 163>, //CLK_TOP_APLL1 + <&topckgen 166>, //CLK_TOP_APLL2 + <&topckgen 233>, //CLK_TOP_APLL12_DIV0 + <&topckgen 234>, //CLK_TOP_APLL12_DIV1 + <&topckgen 235>, //CLK_TOP_APLL12_DIV2 + <&topckgen 236>, //CLK_TOP_APLL12_DIV3 + <&topckgen 238>, //CLK_TOP_APLL12_DIV9 + <&topckgen 100>, //CLK_TOP_A1SYS_HP_SEL + <&topckgen 33>, //CLK_TOP_AUD_INTBUS_SEL + <&topckgen 34>, //CLK_TOP_AUDIO_H_SEL + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS_SEL + <&topckgen 98>, //CLK_TOP_DPTX_M_SEL + <&topckgen 94>, //CLK_TOP_I2SO1_M_SEL + <&topckgen 95>, //CLK_TOP_I2SO2_M_SEL + <&topckgen 96>, //CLK_TOP_I2SI1_M_SEL + <&topckgen 97>, //CLK_TOP_I2SI2_M_SEL + <&infracfg_ao 50>, //CLK_INFRA_AO_AUDIO_26M_B + <&scp_adsp 0>; //CLK_SCP_ADSP_AUDIODSP + clock-names = "clk26m", + "apll1_ck", + "apll2_ck", + "apll12_div0", + "apll12_div1", + "apll12_div2", + "apll12_div3", + "apll12_div9", + "a1sys_hp_sel", + "aud_intbus_sel", + "audio_h_sel", + "audio_local_bus_sel", + "dptx_m_sel", + "i2so1_m_sel", + "i2so2_m_sel", + "i2si1_m_sel", + "i2si2_m_sel", + "infra_ao_audio_26m_b", + "scp_adsp_audiodsp"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml new file mode 100644 index 000000000..356e1feee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8195-mt6359.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/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. + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + - mediatek,mt8195_mt6359_rt1019_rt5682 + - mediatek,mt8195_mt6359_rt1011_rt5682 + - mediatek,mt8195_mt6359_max98390_rt5682 + - mediatek,mt8195_mt6359 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + 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. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - Ext Spk + - Headphone + - IN1P + - Left Spk + - Right Spk + + # Sources + - Headphone L + - Headphone R + - Headset Mic + - HPOL + - HPOR + - Left BE_OUT + - Left SPO + - Right BE_OUT + - Right SPO + - Speaker + + 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. + deprecated: true + + mediatek,hdmi-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8195 HDMI codec node. + deprecated: true + + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8195 ADSP platform. + + mediatek,dai-link: + $ref: /schemas/types.yaml#/definitions/string-array + description: + A list of the desired dai-links in the sound card. Each entry is a + name defined in the machine driver. + +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - DL_SRC_BE + - 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 + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +additionalProperties: false + +required: + - compatible + - mediatek,platform + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + mediatek,dptx-codec: false + mediatek,hdmi-codec: false + +examples: + - | + + sound: mt8195-sound { + compatible = "mediatek,mt8195_mt6359_rt1019_rt5682"; + model = "mt8195_r1019_5682"; + mediatek,platform = <&afe>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + + audio-routing = + "Headphone", "HPOL", + "Headphone", "HPOR", + "IN1P", "Headset Mic", + "Ext Spk", "Speaker"; + + mm-dai-link { + link-name = "ETDM1_IN_BE"; + mediatek,clk-provider = "cpu"; + }; + + hs-playback-dai-link { + link-name = "ETDM1_OUT_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&headset_codec>; + }; + }; + + hs-capture-dai-link { + link-name = "ETDM2_IN_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&headset_codec>; + }; + }; + + spk-playback-dai-link { + link-name = "ETDM2_OUT_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&spk_amplifier>; + }; + }; + + hdmi-dai-link { + link-name = "ETDM3_OUT_BE"; + codec { + sound-dai = <&hdmi_tx>; + }; + }; + + displayport-dai-link { + link-name = "DPTX_BE"; + codec { + sound-dai = <&dp_tx>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 000000000..4f5dec5cb --- /dev/null +++ b/Documentation/devicetree/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/Documentation/devicetree/bindings/sound/neofidelity,ntp8835.yaml b/Documentation/devicetree/bindings/sound/neofidelity,ntp8835.yaml new file mode 100644 index 000000000..44d72a2dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/neofidelity,ntp8835.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/neofidelity,ntp8835.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NeoFidelity NTP8835/NTP8835C Amplifiers + +maintainers: + - Igor Prusov <ivprusov@salutedevices.com> + +description: | + The NTP8835 is a single chip full digital audio amplifier + including power stages for stereo amplifier systems. + NTP8835 is integrated with versatile digital audio signal + processing functions, high-performance, high-fidelity fully + digital PWM modulator and two high-power full-bridge MOSFET + power stages. NTP8835C has identical programming interface, + but has different output signal characteristics. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - neofidelity,ntp8835 + - neofidelity,ntp8835c + + reg: + enum: + - 0x2a + - 0x2b + - 0x2c + - 0x2d + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + clocks: + maxItems: 4 + + clock-names: + items: + - const: wck + - const: bck + - const: scl + - const: mclk + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@2b { + compatible = "neofidelity,ntp8835"; + #sound-dai-cells = <0>; + reg = <0x2b>; + reset-gpios = <&gpio 5 GPIO_ACTIVE_LOW>; + clocks = <&clkc 551>, <&clkc 552>, <&clkc 553>, <&clkc 554>; + clock-names = "wck", "bck", "scl", "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/neofidelity,ntp8918.yaml b/Documentation/devicetree/bindings/sound/neofidelity,ntp8918.yaml new file mode 100644 index 000000000..99b739595 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/neofidelity,ntp8918.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/neofidelity,ntp8918.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NeoFidelity NTP8918 Amplifier + +maintainers: + - Igor Prusov <ivprusov@salutedevices.com> + +description: + The NTP8918 is a single chip full digital audio amplifier + including power stage for stereo amplifier system. + The NTP8918 is integrated with versatile digital audio signal + processing functions, high-performance, high-fidelity fully + digital PWM modulator and two high-power full-bridge MOSFET + power stages. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - neofidelity,ntp8918 + + reg: + enum: + - 0x2a + - 0x2b + - 0x2c + - 0x2d + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + clocks: + maxItems: 3 + + clock-names: + items: + - const: wck + - const: scl + - const: bck + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@2a { + compatible = "neofidelity,ntp8918"; + #sound-dai-cells = <0>; + reg = <0x2a>; + clocks = <&clkc 150>, <&clkc 151>, <&clkc 152>; + clock-names = "wck", "scl", "bck"; + reset-gpios = <&gpio 5 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nokia,rx51.txt b/Documentation/devicetree/bindings/sound/nokia,rx51.txt new file mode 100644 index 000000000..72f93d996 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nokia,rx51.txt @@ -0,0 +1,27 @@ +* Nokia N900 audio setup + +Required properties: +- compatible: Should contain "nokia,n900-audio" +- nokia,cpu-dai: phandle for the McBSP node +- nokia,audio-codec: phandles for the main TLV320AIC3X node and the + auxiliary TLV320AIC3X node (in this order) +- nokia,headphone-amplifier: phandle for the TPA6130A2 node +- tvout-selection-gpios: GPIO for tvout selection +- jack-detection-gpios: GPIO for jack detection +- eci-switch-gpios: GPIO for ECI (Enhancement Control Interface) switch +- speaker-amplifier-gpios: GPIO for speaker amplifier + +Example: + +sound { + compatible = "nokia,n900-audio"; + + nokia,cpu-dai = <&mcbsp2>; + nokia,audio-codec = <&tlv320aic3x>, <&tlv320aic3x_aux>; + nokia,headphone-amplifier = <&tpa6130a2>; + + tvout-selection-gpios = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */ + jack-detection-gpios = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */ + eci-switch-gpios = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */ + speaker-amplifier-gpios = <&twl_gpio 7 GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml new file mode 100644 index 000000000..24006e9dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8315.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8315/NAU8318 Mono Class-D Amplifier + +maintainers: + - David Lin <CTLIN0@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8315 + - nuvoton,nau8318 + + '#sound-dai-cells': + const: 0 + + enable-gpios: + maxItems: 1 + description: + GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + codec { + compatible = "nuvoton,nau8315"; + #sound-dai-cells = <0>; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml new file mode 100644 index 000000000..979be0d33 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8325 audio Amplifier + +maintainers: + - Seven Lee <WTLI@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8325 + + reg: + maxItems: 1 + + nuvoton,vref-impedance-ohms: + description: + The vref impedance to be used in ohms. Middle of voltage enables + Tie-Off selection options. Due to the high impedance of the VREF + pin, it is important to use a low-leakage capacitor. + + enum: [0, 25000, 125000, 2500] + + nuvoton,dac-vref-microvolt: + description: + The DAC vref to be used in voltage. DAC reference voltage setting. Can + be used for minor tuning of the output level. Since the VDDA is range + between 1.62 to 1.98 voltage, the typical value for design is 1.8V. After + the minor tuning, the final microvolt are as the below. + + enum: [1800000, 2700000, 2880000, 3060000] + + nuvoton,alc-enable: + description: + Enable digital automatic level control (ALC) function. + type: boolean + + nuvoton,clock-detection-disable: + description: + When clock detection is enabled, it will detect whether MCLK + and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz. + FS range is from 8kHz to 96kHz. And also needs to detect the ratio + MCLK_SRC/LRCK of 256, 400 or 500, and needs to detect the BCLK + to make sure data is present. There needs to be at least 8 BCLK + cycles per Frame Sync. + type: boolean + + nuvoton,clock-det-data: + description: + Request clock detection to require 2048 non-zero samples before enabling + the audio paths. If set then non-zero samples is required, otherwise it + doesn't matter. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@21 { + compatible = "nuvoton,nau8325"; + reg = <0x21>; + nuvoton,vref-impedance-ohms = <125000>; + nuvoton,dac-vref-microvolt = <2880000>; + nuvoton,alc-enable; + nuvoton,clock-det-data; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000..9f44168ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.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/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,key-enable: + description: handles key press detection. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + nuvoton,dmic-slew-rate: + description: The range 0 to 7 represents the speed of DMIC slew rate. + The lowest value 0 means the slowest rate and the highest value + 7 means the fastest rate. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + + nuvoton,adc-delay-ms: + description: Delay (in ms) to make input path stable and avoid pop noise. + minimum: 125 + maximum: 500 + default: 125 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,key-enable; + nuvoton,left-input-single-end; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + nuvoton,dmic-slew-rate = <0>; + nuvoton,adc-delay-ms = <125>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml new file mode 100644 index 000000000..cf4c13038 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + vdda-supply: + description: Analog power supply + + vddb-supply: + description: Digital buffer (input/output) supply + + vddc-supply: + description: Digital core supply + + vddspk-supply: + description: + Speaker supply (power supply pin for RSPKOUT, LSPKOUT, AUXOUT2 and + AUXTOUT1 output drivers) + + 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: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; + vdda-supply = <&vcca_3v3_s0>; + vddb-supply = <&vcca_3v3_s0>; + vddc-supply = <&vcca_3v3_s0>; + vddspk-supply = <&vcca_3v3_s0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml new file mode 100644 index 000000000..232dc16a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml @@ -0,0 +1,190 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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 + + clocks: + items: + - description: The phandle of the master clock to the CODEC + + clock-names: + items: + - const: mclk + + interrupts: + maxItems: 1 + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 8 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 8 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500ns + - 1 # 1us + - 2 # 2us + - 3 # 4us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2us + - 1 # 4us + - 2 # 8us + - 3 # 16us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + default: 0 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Jack ejection debounce time. + enum: + - 0 # 0 ms + - 1 # 1 ms + - 2 # 10 ms + default: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,sar-threshold-num = <4>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml new file mode 100644 index 000000000..4ebbcb4e6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml @@ -0,0 +1,253 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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> + +description: | + NAU8825 audio CODEC + + Pins on the device (for linking into audio routes): + + Outputs: + * HPOL : Headphone Left Output + * HPOR : Headphone Right Output + * MICBIAS : Microphone Bias Output + + Inputs: + * MICP : Analog Microphone Positive Input + * MICN : Analog Microphone Negative Input + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8825 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: + Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: + Enable JKDET pin pull. + If set - pin pull enabled, otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: + Pull-up JKDET pin. + If set then JKDET pin is pull up, otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500 ns + - 1 # 1 us + - 2 # 2 us + - 3 # 4 us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2 us + - 1 # 4 us + - 2 # 8 us + - 3 # 16 us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + - 3 # 30 ms + default: 3 + + nuvoton,jack-insert-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms. + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms + maximum: 7 + default: 0 + + nuvoton,crosstalk-enable: + description: + make crosstalk function enable if set. + type: boolean + + nuvoton,adcout-drive-strong: + description: + make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + type: boolean + + nuvoton,adc-delay-ms: + description: + Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_pmc 1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml new file mode 100644 index 000000000..96f2f927a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-alc5632.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with ALC5632 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-alc5632(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-alc5632 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headset Stereophone + - Int Spk + - Headset Mic + - Digital Mic + + # CODEC Pins + - SPKOUT + - SPKOUTN + - MICBIAS1 + - MIC1 + - HPR + - HPL + - DMICDAT + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "Headset Mic", "MICBIAS1", + "MIC1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL", + "DMICDAT", "Digital Mic"; + + nvidia,i2s-controller = <&i2s>; + nvidia,audio-codec = <&codec>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml new file mode 100644 index 000000000..2588589ad --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Common properties for NVIDIA Tegra audio complexes + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +properties: + clocks: + items: + - description: PLL A clock + - description: PLL A OUT0 clock + - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk + + clock-names: + items: + - const: pll_a + - const: pll_a_out0 + - const: mclk + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + + nvidia,ac97-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the AC97 controller + + nvidia,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the Tegra I2S controller + + nvidia,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,ear-sel-gpios: + maxItems: 1 + description: The GPIO that switch between the microphones + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + + nvidia,coupled-mic-hp-det: + type: boolean + description: The Mic detect GPIO is viable only if HP detect GPIO is active + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-cpcap.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-cpcap.yaml new file mode 100644 index 000000000..69af2022d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-cpcap.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-cpcap.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with CPCAP CODEC + +maintainers: + - Svyatoslav Ryhel <clamor95@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^motorola,tegra-audio-cpcap(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-cpcap + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names for sources and sinks are + the pins (documented in the binding document), and the jacks on the + board. + minItems: 2 + items: + enum: + # Board Connectors + - Speakers + - Int Spk + - Earpiece + - Int Mic + - Headset Mic + - Internal Mic 1 + - Internal Mic 2 + - Headphone + - Headphones + - Headphone Jack + - Mic Jack + + # CODEC Pins + - MICR + - HSMIC + - EMUMIC + - MICL + - EXTR + - EXTL + - EP + - SPKR + - SPKL + - LINER + - LINEL + - HSR + - HSL + - EMUR + - EMUL + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra20-car.h> + #include <dt-bindings/soc/tegra-pmc.h> + sound { + compatible = "motorola,tegra-audio-cpcap-olympus", + "nvidia,tegra-audio-cpcap"; + nvidia,model = "Motorola Atrix 4G (MB860) CPCAP"; + + nvidia,audio-routing = + "Headphones", "HSR", + "Headphones", "HSL", + "Int Spk", "SPKR", + "Int Spk", "SPKL", + "Earpiece", "EP", + "HSMIC", "Mic Jack", + "MICR", "Internal Mic 1", + "MICL", "Internal Mic 2"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&cpcap_audio>; + + clocks = <&tegra_car TEGRA20_CLK_PLL_A>, + <&tegra_car TEGRA20_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA20_CLK_CDEV1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml new file mode 100644 index 000000000..92bc3ef56 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + - nvidia,tegra238-audio-graph-card + - nvidia,tegra264-audio-graph-card + + clocks: + minItems: 2 + + clock-names: + items: + - const: pll_a + - const: plla_out0 + + assigned-clocks: + minItems: 1 + maxItems: 3 + + assigned-clock-parents: + minItems: 1 + maxItems: 3 + + assigned-clock-rates: + minItems: 1 + maxItems: 3 + + interconnects: + items: + - description: APE read memory client + - description: APE write memory client + + interconnect-names: + items: + - const: dma-mem # read + - const: write + + iommus: + maxItems: 1 + +required: + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +unevaluatedProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + tegra_sound { + compatible = "nvidia,tegra210-audio-graph-card"; + + clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + clock-names = "pll_a", "plla_out0"; + + assigned-clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA210_CLK_EXTERN1>; + assigned-clock-parents = <0>, <0>, <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <368640000>, <49152000>, <12288000>; + + dais = /* FE */ + <&admaif1_port>, + /* Router */ + <&xbar_i2s1_port>, + /* I/O DAP Ports */ + <&i2s1_port>; + + label = "jetson-tx1-ape"; + }; + + // The ports are defined for AHUB and its child devices. + ahub@702d0800 { + compatible = "nvidia,tegra210-ahub"; + reg = <0x702d0800 0x800>; + clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + clock-names = "ahub"; + assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0x702d0000 0x702d0000 0x0000e400>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + xbar_admaif1_ep: endpoint { + remote-endpoint = <&admaif1_ep>; + }; + }; + + // ... + + xbar_i2s1_port: port@a { + reg = <0xa>; + xbar_i2s1_ep: endpoint { + remote-endpoint = <&i2s1_cif_ep>; + }; + }; + }; + + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + admaif1_port: port@0 { + reg = <0x0>; + admaif1_ep: endpoint { + remote-endpoint = <&xbar_admaif1_ep>; + }; + }; + + // More ADMAIF ports to follow + }; + }; + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + reg = <0x702d1000 0x100>; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + + i2s1_cif_ep: endpoint { + remote-endpoint = <&xbar_i2s1_ep>; + }; + }; + + i2s1_port: port@1 { + reg = <0x1>; + + i2s1_dap: endpoint { + dai-format = "i2s"; + }; + }; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml new file mode 100644 index 000000000..4957645a8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml @@ -0,0 +1,95 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Speakers + - Int Spk + - Headphone + - Headphones + - Headphone Jack + - Earpiece + - Headset Mic + - Mic Jack + - Int Mic + - Internal Mic 1 + - Internal Mic 2 + + # CODEC Pins + - HPL + - HPR + - SPKL + - SPKR + - RECL + - RECR + - INA1 + - INA2 + - INB1 + - INB2 + - MIC1 + - MIC2 + - MICBIAS + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + #include <dt-bindings/soc/tegra-pmc.h> + sound { + compatible = "lg,tegra-audio-max98089-p895", + "nvidia,tegra-audio-max98089"; + nvidia,model = "LG Optimus Vu MAX98089"; + + nvidia,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Int Spk", "SPKL", + "Int Spk", "SPKR", + "Earpiece", "RECL", + "Earpiece", "RECR", + "INA1", "Headset Mic", + "MIC1", "MICBIAS", + "MICBIAS", "Internal Mic 1", + "MIC2", "Internal Mic 2"; + + nvidia,i2s-controller = <&tegra_i2s0>; + nvidia,audio-codec = <&codec>; + + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_pmc TEGRA_PMC_CLK_OUT_1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml new file mode 100644 index 000000000..4d912458b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml @@ -0,0 +1,97 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with MAX98090 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98090(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-max98090 + - items: + - enum: + - nvidia,tegra-audio-max98090-nyan-big + - nvidia,tegra-audio-max98090-nyan-blaze + - const: nvidia,tegra-audio-max98090-nyan + - const: nvidia,tegra-audio-max98090 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphones + - Speakers + - Mic Jack + - Int Mic + + # CODEC Pins + - MIC1 + - MIC2 + - DMICL + - DMICR + - IN1 + - IN2 + - IN3 + - IN4 + - IN5 + - IN6 + - IN12 + - IN34 + - IN56 + - HPL + - HPR + - SPKL + - SPKR + - RCVL + - RCVR + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra124-car.h> + + sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml new file mode 100644 index 000000000..0c8067c3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5631.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5631 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5631(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5631 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Int Spk + - Headphone Jack + - Mic Jack + - Int Mic + + # CODEC Pins + - MIC1 + - MIC2 + - AXIL + - AXIR + - MONOIN_RXN + - MONOIN_RXP + - DMIC + - MIC Bias1 + - MIC Bias2 + - MONO_IN + - AUXO1 + - AUXO2 + - SPOL + - SPOR + - HPOL + - HPOR + - MONO + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + #include <dt-bindings/soc/tegra-pmc.h> + sound { + compatible = "asus,tegra-audio-rt5631-tf700t", + "nvidia,tegra-audio-rt5631"; + nvidia,model = "Asus Transformer Infinity TF700T RT5631"; + + nvidia,audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR", + "Int Spk", "SPOL", + "Int Spk", "SPOR", + "MIC1", "MIC Bias1", + "MIC Bias1", "Mic Jack", + "DMIC", "Int Mic"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5631>; + + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_pmc TEGRA_PMC_CLK_OUT_1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml new file mode 100644 index 000000000..263859243 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5640.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5639 or RT5640 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt56(39|40)(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5640 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphones + - Speakers + - Mic Jack + + # CODEC Pins + - DMIC1 + - DMIC2 + - MICBIAS1 + - IN1P + - IN1R + - IN2P + - IN2R + - HPOL + - HPOR + - LOUTL + - LOUTR + - MONOP + - MONON + - SPOLP + - SPOLN + - SPORP + - SPORN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5640-dalmore", + "nvidia,tegra-audio-rt5640"; + nvidia,model = "NVIDIA Tegra Dalmore"; + + nvidia,audio-routing = + "Headphones", "HPOR", + "Headphones", "HPOL", + "Speakers", "SPORP", + "Speakers", "SPORN", + "Speakers", "SPOLP", + "Speakers", "SPOLN"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5640>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, <&clk 217>, <&clk 120>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml new file mode 100644 index 000000000..09e1d0b18 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5677.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5677 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5677(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5677 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphone + - Speaker + - Headset Mic + - Internal Mic 1 + - Internal Mic 2 + + # CODEC Pins + - IN1P + - IN1N + - IN2P + - IN2N + - MICBIAS1 + - DMIC1 + - DMIC2 + - DMIC3 + - DMIC4 + - DMIC L1 + - DMIC L2 + - DMIC L3 + - DMIC L4 + - DMIC R1 + - DMIC R2 + - DMIC R3 + - DMIC R4 + - LOUT1 + - LOUT2 + - LOUT3 + - PDM1L + - PDM1R + - PDM2L + - PDM2R + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, + <&clk 217>, + <&clk 121>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml new file mode 100644 index 000000000..e5bc6a6ad --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with SGTL5000 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-sgtl5000([-_][a-z0-9]+)+$' + - const: nvidia,tegra-audio-sgtl5000 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + - Mic Jack + + # CODEC Pins + - HP_OUT + - LINE_OUT + - LINE_IN + - MIC_IN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30 SGTL5000"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml new file mode 100644 index 000000000..8c87cd166 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-trimslice.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with TrimSlice CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + const: nvidia,tegra-audio-trimslice + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-trimslice"; + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml new file mode 100644 index 000000000..3323d6a43 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8753.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8753 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8753(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8753 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphone Jack + - Mic Jack + + # CODEC Pins + - LOUT1 + - LOUT2 + - ROUT1 + - ROUT2 + - MONO1 + - MONO2 + - OUT3 + - OUT4 + - LINE1 + - LINE2 + - RXP + - RXN + - ACIN + - ACOP + - MIC1N + - MIC1 + - MIC2N + - MIC2 + - Mic Bias + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8753-whistler", + "nvidia,tegra-audio-wm8753"; + nvidia,model = "tegra-wm8753-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "LOUT1", + "Headphone Jack", "ROUT1"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8753>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml new file mode 100644 index 000000000..1be25ce45 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8903.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8903 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8903(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8903 + - items: + - pattern: ad,tegra-audio-plutux + - const: nvidia,tegra-audio-wm8903 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphone Jack + - Int Spk + - Mic Jack + - Int Mic + + # CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - DMICDAT + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - LOP + - LON + - ROP + - RON + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8903-harmony", + "nvidia,tegra-audio-wm8903"; + nvidia,model = "tegra-wm8903-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "HPOUTR", + "Headphone Jack", "HPOUTL", + "Int Spk", "ROP", + "Int Spk", "RON", + "Int Spk", "LOP", + "Int Spk", "LON", + "Mic Jack", "MICBIAS", + "IN1L", "Mic Jack"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8903>; + + nvidia,spkr-en-gpios = <&codec 2 0>; + nvidia,hp-det-gpios = <&gpio 178 0>; + nvidia,int-mic-en-gpios = <&gpio 184 0>; + nvidia,ext-mic-en-gpios = <&gpio 185 0>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8962.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8962.yaml new file mode 100644 index 000000000..2c3bf5a02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8962.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8962.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8962 CODEC + +maintainers: + - Svyatoslav Ryhel <clamor95@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8962(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8962 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names for sources and sinks are + the pins (documented in the binding document), and the jacks on the + board. + minItems: 2 + items: + enum: + # Board Connectors + - Speakers + - Int Spk + - Earpiece + - Int Mic + - Headset Mic + - Internal Mic 1 + - Internal Mic 2 + - Headphone + - Headphones + - Headphone Jack + - Mic Jack + + # CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - IN4L + - IN4R + - DMICDAT + - HPOUTL + - HPOUTR + - SPKOUT + - SPKOUTL + - SPKOUTR + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + #include <dt-bindings/soc/tegra-pmc.h> + sound { + compatible = "microsoft,tegra-audio-wm8962-surface-rt", + "nvidia,tegra-audio-wm8962"; + nvidia,model = "Microsoft Surface RT WM8962"; + + nvidia,audio-routing = + "Headphone Jack", "HPOUTR", + "Headphone Jack", "HPOUTL", + "Int Spk", "SPKOUTR", + "Int Spk", "SPKOUTL"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&wm8962>; + + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_pmc TEGRA_PMC_CLK_OUT_1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml new file mode 100644 index 000000000..397306b88 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm9712.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM9712 CODEC + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Thierry Reding <thierry.reding@gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm9712([-_][a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm9712 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - Headphone + - LineIn + - Mic + + # CODEC Pins + - MONOOUT + - HPOUTL + - HPOUTR + - LOUT2 + - ROUT2 + - OUT3 + - LINEINL + - LINEINR + - PHONE + - PCBEEP + - MIC1 + - MIC2 + - Mic Bias + +required: + - nvidia,ac97-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm9712-colibri_t20", + "nvidia,tegra-audio-wm9712"; + nvidia,model = "Toradex Colibri T20"; + + nvidia,audio-routing = + "Headphone", "HPOUTL", + "Headphone", "HPOUTR", + "LineIn", "LINEINL", + "LineIn", "LINEINR", + "Mic", "MIC1"; + + nvidia,ac97-controller = <&ac97>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml new file mode 100644 index 000000000..66b56e715 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,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: + - enum: + - nvidia,tegra186-asrc + - nvidia,tegra264-asrc + - items: + - enum: + - nvidia,tegra234-asrc + - nvidia,tegra194-asrc + - const: nvidia,tegra186-asrc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^ASRC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + ASRC has seven input ports and six output ports. Accordingly ACIF + (Audio Client Interfaces) port nodes are defined to represent the + ASRC inputs (port 0 to 6) and outputs (port 7 to 12). These are + connected to corresponding ports on AHUB (Audio Hub). Additional + input (port 6) is for receiving ratio information from estimator. + + patternProperties: + '^port@[0-6]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF input ports + '^port@[7-9]|1[1-2]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ASRC ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + asrc@2910000 { + compatible = "nvidia,tegra186-asrc"; + reg = <0x2910000 0x2000>; + sound-name-prefix = "ASRC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml new file mode 100644 index 000000000..46ba16708 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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,tegra264-dspk + - nvidia,tegra234-dspk + - nvidia,tegra194-dspk + - const: nvidia,tegra186-dspk + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: dspk + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + sound-name-prefix: + pattern: "^DSPK[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DSPK ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DSPK DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + - sound-name-prefix + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra186-clock.h> + + dspk@2905000 { + compatible = "nvidia,tegra186-dspk"; + reg = <0x2905000 0x100>; + clocks = <&bpmp TEGRA186_CLK_DSPK1>; + clock-names = "dspk"; + assigned-clocks = <&bpmp TEGRA186_CLK_DSPK1>; + assigned-clock-parents = <&bpmp TEGRA186_CLK_PLL_A_OUT0>; + assigned-clock-rates = <12288000>; + sound-name-prefix = "DSPK1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml new file mode 100644 index 000000000..4ea0a303d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-ac97.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 AC97 controller + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-ac97 + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: ac97 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + dmas: + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,codec-reset-gpios: + description: Reset pin of external AC97 codec + maxItems: 1 + + nvidia,codec-sync-gpios: + description: AC97 DAP _FS line + maxItems: 1 + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + - nvidia,codec-reset-gpios + - nvidia,codec-sync-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra20-car.h> + #include <dt-bindings/gpio/tegra-gpio.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/gpio/gpio.h> + + ac97@70002000 { + compatible = "nvidia,tegra20-ac97"; + reg = <0x70002000 0x200>; + resets = <&tegra_car 3>; + reset-names = "ac97"; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car 3>; + dmas = <&apbdma 12>, <&apbdma 12>; + dma-names = "rx", "tx"; + nvidia,codec-reset-gpios = <&gpio TEGRA_GPIO(V, 2) GPIO_ACTIVE_HIGH>; + nvidia,codec-sync-gpios = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml new file mode 100644 index 000000000..44c5ce8ee --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-das.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra 20 DAS (Digital Audio Switch) controller + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-das + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + bus { + #address-cells = <1>; + #size-cells = <1>; + das@70000c00 { + compatible = "nvidia,tegra20-das"; + reg = <0x70000c00 0x80>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml new file mode 100644 index 000000000..68ae124ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 I2S Controller + +description: | + The I2S Controller streams synchronous serial audio data between system + memory and an external audio device. The controller supports the I2S Left + Justified Mode, Right Justified Mode, and DSP mode formats. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-i2s + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: i2s + + interrupts: + maxItems: 1 + + clocks: + minItems: 1 + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = <45>; + clocks = <&tegra_car 11>; + resets = <&tegra_car 11>; + reset-names = "i2s"; + dmas = <&apbdma 21>, <&apbdma 21>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml new file mode 100644 index 000000000..dc76a4dc0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 S/PDIF Controller + +description: | + The S/PDIF controller supports both input and output in serial audio + digital interface format. The input controller can digitally recover + a clock from the received stream. The S/PDIF controller is also used + to generate the embedded audio for HDMI output channel. + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nvidia,tegra20-spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 2 + + clock-names: + items: + - const: out + - const: in + + dmas: + minItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + + nvidia,fixed-parent-rate: + description: | + Specifies whether board prefers parent clock to stay at a fixed rate. + This allows multiple Tegra20 audio components work simultaneously by + limiting number of supportable audio rates. + type: boolean + +required: + - compatible + - reg + - resets + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + spdif@70002400 { + compatible = "nvidia,tegra20-spdif"; + reg = <0x70002400 0x200>; + interrupts = <77>; + clocks = <&clk 99>, <&clk 98>; + clock-names = "out", "in"; + resets = <&rst 10>; + dmas = <&apbdma 3>, <&apbdma 3>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml new file mode 100644 index 000000000..2ce4049f9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml @@ -0,0 +1,172 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 + - nvidia,tegra264-admaif + - items: + - enum: + - nvidia,tegra234-admaif + - nvidia,tegra194-admaif + - const: nvidia,tegra186-admaif + + reg: + maxItems: 1 + + dmas: true + + dma-names: true + + interconnects: + items: + - description: APE read memory client + - description: APE write memory client + + interconnect-names: + items: + - const: dma-mem # read + - const: write + + iommus: + maxItems: 1 + + 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]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: nvidia,tegra210-admaif + then: + properties: + dmas: + description: + DMA channel specifiers, equally divided for Tx and Rx. + minItems: 1 + maxItems: 20 + dma-names: + items: + pattern: "^[rt]x(10|[1-9])$" + description: + Should be "rx1", "rx2" ... "rx10" for DMA Rx channel + Should be "tx1", "tx2" ... "tx10" for DMA Tx channel + minItems: 1 + maxItems: 20 + interconnects: false + interconnect-names: false + iommus: false + + - if: + properties: + compatible: + contains: + const: nvidia,tegra186-admaif + then: + 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)$" + description: + Should be "rx1", "rx2" ... "rx20" for DMA Rx channel + Should be "tx1", "tx2" ... "tx20" for DMA Tx channel + minItems: 1 + maxItems: 40 + + - if: + properties: + compatible: + contains: + const: nvidia,tegra264-admaif + then: + properties: + dmas: + description: + DMA channel specifiers, equally divided for Tx and Rx. + minItems: 1 + maxItems: 64 + dma-names: + items: + pattern: "^[rt]x(3[0-2]|[1-2][0-9]|[1-9])$" + description: + Should be "rx1", "rx2" ... "rx32" for DMA Rx channel + Should be "tx1", "tx2" ... "tx32" for DMA Tx channel + minItems: 1 + maxItems: 64 + +required: + - compatible + - reg + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml new file mode 100644 index 000000000..19a80929f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-adx.yaml# +$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: + - enum: + - nvidia,tegra210-adx + - nvidia,tegra264-adx + - items: + - enum: + - nvidia,tegra234-adx + - nvidia,tegra194-adx + - nvidia,tegra186-adx + - const: nvidia,tegra210-adx + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^ADX[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + ADX has one input and four outputs. Accordingly ACIF (Audio Client + Interface) port nodes are defined to represent ADX input (port 0) + and outputs (ports 1 to 4). These are connected to corresponding + ports on AHUB (Audio Hub). + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF input port + patternProperties: + '^port@[1-4]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: ADX ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + adx@702d3800 { + compatible = "nvidia,tegra210-adx"; + reg = <0x702d3800 0x100>; + sound-name-prefix = "ADX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml new file mode 100644 index 000000000..1c9f24d26 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml @@ -0,0 +1,197 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 + - nvidia,tegra264-ahub + - items: + - const: nvidia,tegra194-ahub + - const: nvidia,tegra186-ahub + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: ahub + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + "#address-cells": + enum: [ 1, 2 ] + + "#size-cells": + enum: [ 1, 2 ] + + ranges: true + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Contains list of ACIF (Audio CIF) port nodes for AHUB (Audio Hub). + These are connected to ACIF interfaces of AHUB clients. Thus the + number of port nodes depend on the number of clients that AHUB may + have depending on the SoC revision. + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +patternProperties: + '^i2s@[0-9a-f]+$': + type: object + + '^dmic@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-dmic.yaml# + + '^admaif@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-admaif.yaml# + + '^dspk@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-dspk.yaml# + + '^mvc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mvc.yaml# + + '^sfc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-sfc.yaml# + + '^amx@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-amx.yaml# + + '^adx@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-adx.yaml# + + '^amixer@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mixer.yaml# + + '^asrc@[0-9a-f]+$': + type: object + $ref: nvidia,tegra186-asrc.yaml# + + '^processing-engine@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-ope.yaml# + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + - "#address-cells" + - "#size-cells" + - ranges + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + ahub@702d0800 { + compatible = "nvidia,tegra210-ahub"; + reg = <0x702d0800 0x800>; + clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + clock-names = "ahub"; + assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0x702d0000 0x702d0000 0x0000e400>; + + // All AHUB child nodes below + admaif@702d0000 { + compatible = "nvidia,tegra210-admaif"; + reg = <0x702d0000 0x800>; + dmas = <&adma 1>, <&adma 1>, + <&adma 2>, <&adma 2>, + <&adma 3>, <&adma 3>, + <&adma 4>, <&adma 4>, + <&adma 5>, <&adma 5>, + <&adma 6>, <&adma 6>, + <&adma 7>, <&adma 7>, + <&adma 8>, <&adma 8>, + <&adma 9>, <&adma 9>, + <&adma 10>, <&adma 10>; + dma-names = "rx1", "tx1", + "rx2", "tx2", + "rx3", "tx3", + "rx4", "tx4", + "rx5", "tx5", + "rx6", "tx6", + "rx7", "tx7", + "rx8", "tx8", + "rx9", "tx9", + "rx10", "tx10"; + }; + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + reg = <0x702d1000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + sound-name-prefix = "I2S1"; + }; + + dmic@702d4000 { + compatible = "nvidia,tegra210-dmic"; + reg = <0x702d4000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + clock-names = "dmic"; + assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <3072000>; + sound-name-prefix = "DMIC1"; + }; + + // More child nodes to follow + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml new file mode 100644 index 000000000..89712102c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.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,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: + - enum: + - nvidia,tegra210-amx + - nvidia,tegra194-amx + - nvidia,tegra264-amx + - items: + - const: nvidia,tegra186-amx + - const: nvidia,tegra210-amx + - items: + - const: nvidia,tegra234-amx + - const: nvidia,tegra194-amx + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^AMX[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + AMX has four inputs and one output. Accordingly ACIF (Audio Client + Interfaces) port nodes are defined to represent AMX inputs (port 0 + to 3) and output (port 4). These are connected to corresponding + ports on AHUB (Audio Hub). + + patternProperties: + '^port@[0-3]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF input ports + + properties: + port@4: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: AMX ACIF output port + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amx@702d3000 { + compatible = "nvidia,tegra210-amx"; + reg = <0x702d3000 0x100>; + sound-name-prefix = "AMX1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml new file mode 100644 index 000000000..bb8088878 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,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,tegra264-dmic + - nvidia,tegra234-dmic + - nvidia,tegra194-dmic + - nvidia,tegra186-dmic + - const: nvidia,tegra210-dmic + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: dmic + + assigned-clocks: + maxItems: 1 + + assigned-clock-parents: + maxItems: 1 + + assigned-clock-rates: + maxItems: 1 + + sound-name-prefix: + pattern: "^DMIC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + DMIC DAP (Digital Audio Port) interface which can be connected + to external audio codec for capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + dmic@702d4000 { + compatible = "nvidia,tegra210-dmic"; + reg = <0x702d4000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + clock-names = "dmic"; + assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <3072000>; + sound-name-prefix = "DMIC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml new file mode 100644 index 000000000..903e815af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml @@ -0,0 +1,117 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + - enum: + - nvidia,tegra210-i2s + - nvidia,tegra264-i2s + - items: + - enum: + - nvidia,tegra234-i2s + - nvidia,tegra194-i2s + - nvidia,tegra186-i2s + - const: nvidia,tegra210-i2s + + reg: + maxItems: 1 + + clocks: + minItems: 1 + items: + - description: I2S bit clock + - description: + Sync input clock, which can act as clock source to other I/O + modules in AHUB. The Tegra I2S driver sets this clock rate as + per bit clock rate. I/O module which wants to use this clock + as source, can mention this clock as parent in the DT bindings. + This is an optional clock entry, since it is only required when + some other I/O wants to reference from a particular I2Sx + instance. + + clock-names: + minItems: 1 + items: + - const: i2s + - const: sync_input + + assigned-clocks: + minItems: 1 + maxItems: 2 + + assigned-clock-parents: + minItems: 1 + maxItems: 2 + + assigned-clock-rates: + minItems: 1 + maxItems: 2 + + sound-name-prefix: + pattern: "^I2S[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S ACIF (Audio Client Interface) port connected to the + corresponding AHUB (Audio Hub) ACIF port. + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + I2S DAP (Digital Audio Port) interface which can be connected + to external audio codec for playback or capture. + +required: + - compatible + - reg + - clocks + - clock-names + - assigned-clocks + - assigned-clock-parents + +additionalProperties: false + +examples: + - | + #include<dt-bindings/clock/tegra210-car.h> + + i2s@702d1000 { + compatible = "nvidia,tegra210-i2s"; + reg = <0x702d1000 0x100>; + clocks = <&tegra_car TEGRA210_CLK_I2S0>; + clock-names = "i2s"; + assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>; + assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <1536000>; + sound-name-prefix = "I2S1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml new file mode 100644 index 000000000..4c121b9cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-mbdrc.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 MBDRC + +description: + The Multi Band Dynamic Range Compressor (MBDRC) is part of Output + Processing Engine (OPE) which interfaces with Audio Hub (AHUB) via + Audio Client Interface (ACIF). MBDRC can be used as a traditional + single full band or a dual band or a multi band dynamic processor. + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-mbdrc + - items: + - enum: + - nvidia,tegra264-mbdrc + - nvidia,tegra234-mbdrc + - nvidia,tegra194-mbdrc + - nvidia,tegra186-mbdrc + - const: nvidia,tegra210-mbdrc + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + dynamic-range-compressor@702d8200 { + compatible = "nvidia,tegra210-mbdrc"; + reg = <0x702d8200 0x200>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml new file mode 100644 index 000000000..56b4c4fc1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,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,tegra264-amixer + - nvidia,tegra234-amixer + - nvidia,tegra194-amixer + - nvidia,tegra186-amixer + - const: nvidia,tegra210-amixer + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^MIXER[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + description: | + Mixer has ten inputs and five outputs. Accordingly ACIF (Audio + Client Interfaces) port nodes are defined to represent Mixer + inputs (port 0 to 9) and outputs (port 10 to 14). These are + connected to corresponding ports on AHUB (Audio Hub). + + patternProperties: + '^port@[0-9]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF input ports + '^port@[10-14]': + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Mixer ACIF output ports + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + amixer@702dbb00 { + compatible = "nvidia,tegra210-amixer"; + reg = <0x702dbb00 0x800>; + sound-name-prefix = "MIXER1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml new file mode 100644 index 000000000..bde4ac631 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml @@ -0,0 +1,78 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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,tegra264-mvc + - nvidia,tegra234-mvc + - nvidia,tegra194-mvc + - nvidia,tegra186-mvc + - const: nvidia,tegra210-mvc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^MVC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + MVC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + mvc@702da000 { + compatible = "nvidia,tegra210-mvc"; + reg = <0x702da000 0x200>; + sound-name-prefix = "MVC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml new file mode 100644 index 000000000..756c3096a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,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: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-ope + - items: + - enum: + - nvidia,tegra264-ope + - nvidia,tegra234-ope + - nvidia,tegra194-ope + - nvidia,tegra186-ope + - const: nvidia,tegra210-ope + + reg: + maxItems: 1 + + "#address-cells": + enum: [ 1, 2 ] + + "#size-cells": + enum: [ 1, 2 ] + + ranges: true + + sound-name-prefix: + pattern: "^OPE[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: + OPE ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: + OPE ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +patternProperties: + '^equalizer@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-peq.yaml# + + '^dynamic-range-compressor@[0-9a-f]+$': + type: object + $ref: nvidia,tegra210-mbdrc.yaml# + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + processing-engine@702d8000 { + compatible = "nvidia,tegra210-ope"; + reg = <0x702d8000 0x100>; + sound-name-prefix = "OPE1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml new file mode 100644 index 000000000..2f11a484d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-peq.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Tegra210 PEQ + +description: + The Parametric Equalizer (PEQ) is a cascade of biquad filters with + each filter tuned based on certain parameters. It can be used to + equalize the irregularities in the speaker frequency response. + PEQ sits inside Output Processing Engine (OPE) which interfaces + with Audio Hub (AHUB) via Audio Client Interface (ACIF). + +maintainers: + - Jon Hunter <jonathanh@nvidia.com> + - Mohan Kumar <mkumard@nvidia.com> + - Sameer Pujar <spujar@nvidia.com> + +properties: + compatible: + oneOf: + - const: nvidia,tegra210-peq + - items: + - enum: + - nvidia,tegra264-peq + - nvidia,tegra234-peq + - nvidia,tegra194-peq + - nvidia,tegra186-peq + - const: nvidia,tegra210-peq + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + equalizer@702d8100 { + compatible = "nvidia,tegra210-peq"; + reg = <0x702d8100 0x100>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml new file mode 100644 index 000000000..959aa7fff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra210-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,tegra264-sfc + - nvidia,tegra234-sfc + - nvidia,tegra194-sfc + - nvidia,tegra186-sfc + - const: nvidia,tegra210-sfc + + reg: + maxItems: 1 + + sound-name-prefix: + pattern: "^SFC[1-9]$" + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + SFC ACIF (Audio Client Interface) input port. This is connected + to corresponding ACIF output port on AHUB (Audio Hub). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: | + SFC ACIF output port. This is connected to corresponding ACIF + input port on AHUB. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + sfc@702d2000 { + compatible = "nvidia,tegra210-sfc"; + reg = <0x702d2000 0x200>; + sound-name-prefix = "SFC1"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt new file mode 100644 index 000000000..0e9a1895d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -0,0 +1,88 @@ +NVIDIA Tegra30 AHUB (Audio Hub) + +Required properties: +- compatible : For Tegra30, must contain "nvidia,tegra30-ahub". For Tegra114, + must contain "nvidia,tegra114-ahub". For Tegra124, must contain + "nvidia,tegra124-ahub". Otherwise, must contain "nvidia,<chip>-ahub", + plus at least one of the above, where <chip> is tegra132. +- reg : Should contain the register physical address and length for each of + the AHUB's register blocks. + - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks. + - Tegra114 requires an additional entry, for the APBIF2 register block. +- interrupts : Should contain AHUB interrupt +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - d_audio + - apbif +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: + Tegra30 and later: + - d_audio + - apbif + - i2s0 + - i2s1 + - i2s2 + - i2s3 + - i2s4 + - dam0 + - dam1 + - dam2 + - spdif + Tegra114 and later additionally require: + - amx + - adx + Tegra124 and later additionally require: + - amx1 + - adx1 + - afc0 + - afc1 + - afc2 + - afc3 + - afc4 + - afc5 +- ranges : The bus address mapping for the configlink register bus. + Can be empty since the mapping is 1:1. +- dmas : Must contain an entry for each entry in clock-names. + See ../dma/dma.txt for details. +- dma-names : Must include the following entries: + - rx0 .. rx<n> + - tx0 .. tx<n> + ... where n is: + Tegra30: 3 + Tegra114, Tegra124: 9 +- #address-cells : For the configlink bus. Should be <1>; +- #size-cells : For the configlink bus. Should be <1>. + +AHUB client modules need to specify the IDs of their CIFs (Client InterFaces). +For RX CIFs, the numbers indicate the register number within AHUB routing +register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1). +For TX CIFs, the numbers indicate the bit position within the AHUB routing +registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1). + +Example: + +ahub@70080000 { + compatible = "nvidia,tegra30-ahub"; + reg = <0x70080000 0x200 0x70080200 0x100>; + interrupts = < 0 103 0x04 >; + nvidia,dma-request-selector = <&apbdma 1>; + clocks = <&tegra_car 106>, <&tegra_car 107>; + clock-names = "d_audio", "apbif"; + resets = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>, + <&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>, + <&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>, + <&tegra_car 110>, <&tegra_car 10>; + reset-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2", + "i2s3", "i2s4", "dam0", "dam1", "dam2", + "spdif"; + dmas = <&apbdma 1>, <&apbdma 1>; + <&apbdma 2>, <&apbdma 2>; + <&apbdma 3>, <&apbdma 3>; + <&apbdma 4>, <&apbdma 4>; + dma-names = "rx0", "tx0", "rx1", "tx1", "rx2", "tx2", "rx3", "tx3"; + ranges; + #address-cells = <1>; + #size-cells = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml new file mode 100644 index 000000000..8a8767589 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml @@ -0,0 +1,198 @@ +# 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: + - enum: + - nvidia,tegra30-hda + - nvidia,tegra194-hda + - nvidia,tegra234-hda + - nvidia,tegra264-hda + - items: + - enum: + - nvidia,tegra186-hda + - nvidia,tegra210-hda + - nvidia,tegra124-hda + - nvidia,tegra114-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: 1 + maxItems: 3 + + clock-names: + minItems: 1 + maxItems: 3 + + resets: + minItems: 2 + maxItems: 3 + + reset-names: + minItems: 2 + maxItems: 3 + + 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 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - nvidia,tegra30-hda + then: + properties: + clocks: + minItems: 3 + clock-names: + items: + - const: hda + - const: hda2hdmi + - const: hda2codec_2x + resets: + minItems: 3 + reset-names: + items: + - const: hda + - const: hda2hdmi + - const: hda2codec_2x + - if: + properties: + compatible: + contains: + enum: + - nvidia,tegra194-hda + then: + properties: + clocks: + minItems: 3 + clock-names: + items: + - const: hda + - const: hda2hdmi + - const: hda2codec_2x + resets: + maxItems: 2 + reset-names: + items: + - const: hda + - const: hda2hdmi + - if: + properties: + compatible: + contains: + enum: + - nvidia,tegra234-hda + then: + properties: + clocks: + minItems: 2 + maxItems: 2 + clock-names: + items: + - const: hda + - const: hda2codec_2x + resets: + maxItems: 2 + reset-names: + items: + - const: hda + - const: hda2codec_2x + - if: + properties: + compatible: + contains: + enum: + - nvidia,tegra264-hda + then: + properties: + clocks: + maxItems: 1 + clock-names: + items: + - const: hda + resets: + maxItems: 2 + reset-names: + items: + - const: hda + - const: hda2codec_2x + power-domains: false + +examples: + - | + #include<dt-bindings/clock/tegra124-car-common.h> + #include<dt-bindings/interrupt-controller/arm-gic.h> + + hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hda2codec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>, /* hda2hdmi */ + <&tegra_car 111>; /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hda2codec_2x"; + nvidia,model = "jetson-tk1-hda"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml new file mode 100644 index 000000000..89c3c6414 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.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,tegra30-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra30 I2S controller + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + oneOf: + - enum: + - nvidia,tegra124-i2s + - nvidia,tegra30-i2s + - items: + - const: nvidia,tegra114-i2s + - const: nvidia,tegra30-i2s + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: i2s + + resets: + maxItems: 1 + + reset-names: + const: i2s + + nvidia,ahub-cif-ids: + description: list of AHUB CIF IDs + $ref: /schemas/types.yaml#/definitions/uint32-array + items: + - description: rx (playback) + - description: tx (capture) + +required: + - compatible + - reg + - clocks + - resets + - reset-names + - nvidia,ahub-cif-ids + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + i2s@70080300 { + compatible = "nvidia,tegra30-i2s"; + reg = <0x70080300 0x100>; + nvidia,ahub-cif-ids = <4 4>; + clocks = <&tegra_car TEGRA30_CLK_I2S0>; + resets = <&tegra_car 30>; + reset-names = "i2s"; + }; +... diff --git a/Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml b/Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml new file mode 100644 index 000000000..40a0877a8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,lpc3220-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP LPC32XX I2S Controller + +description: + The I2S controller in LPC32XX SoCs, ASoC DAI. + +maintainers: + - J.M.B. Downing <jonathan.downing@nautel.com> + - Piotr Wojtaszczyk <piotr.wojtaszczyk@timesys.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nxp,lpc3220-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: input clock of the peripheral. + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - dmas + - dma-names + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/lpc32xx-clock.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2s@20094000 { + compatible = "nxp,lpc3220-i2s"; + reg = <0x20094000 0x1000>; + interrupts = <22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk LPC32XX_CLK_I2S0>; + dmas = <&dma 0 1>, <&dma 13 1>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml new file mode 100644 index 000000000..fd2415e23 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,tfa989x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP/Goodix TFA989X (TFA1) Audio Amplifiers + +maintainers: + - Stephan Gerhold <stephan@gerhold.net> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nxp,tfa9890 + - nxp,tfa9895 + - nxp,tfa9897 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + rcv-gpios: + description: optional GPIO to be asserted when receiver mode is enabled. + + sound-name-prefix: true + + vddd-supply: + description: regulator phandle for the VDDD power supply. + +if: + not: + properties: + compatible: + const: nxp,tfa9897 +then: + properties: + rcv-gpios: false + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@34 { + compatible = "nxp,tfa9895"; + reg = <0x34>; + sound-name-prefix = "Speaker Left"; + #sound-dai-cells = <0>; + }; + audio-codec@36 { + compatible = "nxp,tfa9895"; + reg = <0x36>; + sound-name-prefix = "Speaker Right"; + #sound-dai-cells = <0>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + speaker_codec_top: audio-codec@34 { + compatible = "nxp,tfa9897"; + reg = <0x34>; + vddd-supply = <&pm8916_l6>; + rcv-gpios = <&msmgpio 50 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&speaker_top_default>; + sound-name-prefix = "Speaker Top"; + #sound-dai-cells = <0>; + }; + + speaker_codec_bottom: audio-codec@36 { + compatible = "nxp,tfa9897"; + reg = <0x36>; + vddd-supply = <&pm8916_l6>; + rcv-gpios = <&msmgpio 111 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&speaker_bottom_default>; + sound-name-prefix = "Speaker Bottom"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 000000000..462b04e82 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack-detection: Need to be present if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt new file mode 100644 index 000000000..ae8bf703c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -0,0 +1,36 @@ +* Texas Instruments OMAP2+ McBSP module + +Required properties: +- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420 + "ti,omap2430-mcbsp" for McBSP on OMAP2430 + "ti,omap3-mcbsp" for McBSP on OMAP3 + "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC +- reg: Register location and size, for OMAP4+ as an array: + <MPU access base address, size>, + <L3 interconnect address, size>; +- reg-names: Array of strings associated with the address space +- interrupts: Interrupt numbers for the McBSP port, as an array in case the + McBSP IP have more interrupt lines: + <OCP compliant irq>, + <TX irq>, + <RX irq>; +- interrupt-names: Array of strings associated with the interrupt numbers +- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) +- ti,hwmods: Name of the hwmod associated to the McBSP port + +Example: + +mcbsp2: mcbsp@49022000 { + compatible = "ti,omap3-mcbsp"; + reg = <0x49022000 0xff>, + <0x49028000 0xff>; + reg-names = "mpu", "sidetone"; + interrupts = <0 17 0x4>, /* OCP compliant interrupt */ + <0 62 0x4>, /* TX interrupt */ + <0 63 0x4>, /* RX interrupt */ + <0 4 0x4>; /* Sidetone */ + interrupt-names = "common", "tx", "rx", "sidetone"; + interrupt-parent = <&intc>; + ti,buffer-size = <1280>; + ti,hwmods = "mcbsp2"; +}; diff --git a/Documentation/devicetree/bindings/sound/option,gtm601.yaml b/Documentation/devicetree/bindings/sound/option,gtm601.yaml new file mode 100644 index 000000000..ff813d97f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/option,gtm601.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/option,gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel@puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - description: Broadmobi BM818 (48Khz stereo) + items: + - const: broadmobi,bm818 + - const: option,gtm601 + - description: GTM601 (8kHz mono) + const: option,gtm601 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + +allOf: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + codec { + compatible = "option,gtm601"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt new file mode 100644 index 000000000..97de66932 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -0,0 +1,23 @@ +PCM3060 audio CODEC + +This driver supports both I2C and SPI. + +Required properties: + +- compatible: "ti,pcm3060" + +- reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + +Examples: + + pcm3060: pcm3060@46 { + compatible = "ti,pcm3060"; + reg = <0x46>; + ti,out-single-ended = "true"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml new file mode 100644 index 000000000..6ad451549 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,apq8016-sbc-sndcard.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm APQ8016 and similar sound cards + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + - Stephan Gerhold <stephan@gerhold.net> + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + + 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 + + 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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml new file mode 100644 index 000000000..3a559bd07 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -0,0 +1,290 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-cpu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. LPASS CPU dai driver + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + - Rohit kumar <quic_rohkumar@quicinc.com> + +description: | + Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist + of MI2S interface for audio data transfer on external codecs. LPASS cpu driver + is a module to configure Low-Power Audio Interface(LPAIF) core registers + across different IP versions. + +properties: + compatible: + enum: + - qcom,lpass-cpu + - qcom,apq8016-lpass-cpu + - qcom,sc7180-lpass-cpu + - qcom,sc7280-lpass-cpu + + reg: + minItems: 1 + maxItems: 6 + description: LPAIF core registers + + reg-names: + minItems: 1 + maxItems: 6 + + clocks: + minItems: 3 + maxItems: 10 + + clock-names: + minItems: 1 + maxItems: 10 + + interrupts: + minItems: 1 + maxItems: 4 + description: LPAIF DMA buffer interrupt + + interrupt-names: + minItems: 1 + maxItems: 4 + + qcom,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle for the audio DSP node + + iommus: + minItems: 2 + maxItems: 3 + description: Phandle to apps_smmu node with sid mask + + power-domains: + maxItems: 1 + + power-domain-names: + maxItems: 1 + + required-opps: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +patternProperties: + "^dai-link@[0-9a-f]+$": + type: object + description: | + LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node + depends on the specific driver providing the functionality and + properties. + properties: + reg: + maxItems: 1 + description: Must be one of the DAI ID + + qcom,playback-sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: list of MI2S data lines for playback + + qcom,capture-sd-lines: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: list of MI2S data lines for capture + + required: + - reg + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - interrupts + - interrupt-names + - '#sound-dai-cells' + +unevaluatedProperties: false + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: qcom,lpass-cpu + + then: + properties: + clocks: + maxItems: 3 + clock-names: + items: + - const: ahbix-clk + - const: mi2s-osr-clk + - const: mi2s-bit-clk + + - if: + properties: + compatible: + contains: + const: qcom,apq8016-lpass-cpu + + then: + properties: + clocks: + minItems: 7 + maxItems: 7 + clock-names: + items: + - const: ahbix-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + - const: mi2s-bit-clk2 + - const: mi2s-bit-clk3 + - const: pcnoc-mport-clk + - const: pcnoc-sway-clk + + - if: + properties: + compatible: + contains: + const: qcom,sc7180-lpass-cpu + + then: + properties: + clocks: + minItems: 6 + maxItems: 6 + clock-names: + items: + - const: pcnoc-sway-clk + - const: audio-core + - const: mclk0 + - const: pcnoc-mport-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + reg: + minItems: 2 + maxItems: 2 + reg-names: + items: + - const: lpass-hdmiif + - const: lpass-lpaif + interrupts: + minItems: 2 + maxItems: 2 + interrupt-names: + items: + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + required: + - iommus + - power-domains + + - if: + properties: + compatible: + contains: + const: qcom,sc7280-lpass-cpu + + then: + properties: + clocks: + minItems: 10 + maxItems: 10 + clock-names: + items: + - const: aon_cc_audio_hm_h + - const: audio_cc_ext_mclk0 + - const: core_cc_sysnoc_mport_core + - const: core_cc_ext_if0_ibit + - const: core_cc_ext_if1_ibit + - const: audio_cc_codec_mem + - const: audio_cc_codec_mem0 + - const: audio_cc_codec_mem1 + - const: audio_cc_codec_mem2 + - const: aon_cc_va_mem0 + reg: + minItems: 6 + maxItems: 6 + reg-names: + items: + - const: lpass-hdmiif + - const: lpass-lpaif + - const: lpass-rxtx-cdc-dma-lpm + - const: lpass-rxtx-lpaif + - const: lpass-va-lpaif + - const: lpass-va-cdc-dma-lpm + interrupts: + minItems: 4 + maxItems: 4 + interrupt-names: + items: + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - const: lpass-irq-vaif + - const: lpass-irq-rxtxif + power-domain-names: + items: + - const: lcx + + required: + - iommus + - power-domains + +examples: + - | + #include <dt-bindings/sound/sc7180-lpass.h> + + soc { + #address-cells = <2>; + #size-cells = <2>; + lpass@62d80000 { + compatible = "qcom,sc7180-lpass-cpu"; + + reg = <0 0x62d87000 0 0x68000>, + <0 0x62f00000 0 0x29000>; + reg-names = "lpass-hdmiif", + "lpass-lpaif"; + iommus = <&apps_smmu 0x1020 0>, + <&apps_smmu 0x1032 0>; + power-domains = <&lpass_hm 0>; + + clocks = <&gcc 131>, + <&lpasscorecc 6>, + <&lpasscorecc 7>, + <&lpasscorecc 10>, + <&lpasscorecc 8>, + <&lpasscorecc 9>; + + clock-names = "pcnoc-sway-clk", "audio-core", + "mclk0", "pcnoc-mport-clk", + "mi2s-bit-clk0", "mi2s-bit-clk1"; + + interrupts = <0 160 1>, + <0 268 1>; + interrupt-names = "lpass-irq-lpaif", + "lpass-irq-hdmi"; + #sound-dai-cells = <1>; + + #address-cells = <1>; + #size-cells = <0>; + /* Optional to set different MI2S SD lines */ + dai-link@0 { + reg = <MI2S_PRIMARY>; + qcom,playback-sd-lines = <1>; + qcom,capture-sd-lines = <0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml new file mode 100644 index 000000000..4988e7ed6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -0,0 +1,158 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-rx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) RX Macro audio codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm6115-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + - qcom,sm8550-lpass-rx-macro + - qcom,sc8280xp-lpass-rx-macro + - items: + - enum: + - qcom,eliza-lpass-rx-macro + - qcom,kaanapali-lpass-rx-macro + - qcom,sm8650-lpass-rx-macro + - qcom,sm8750-lpass-rx-macro + - qcom,x1e80100-lpass-rx-macro + - const: qcom,sm8550-lpass-rx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 3 + maxItems: 5 + + clock-names: + minItems: 3 + maxItems: 5 + + clock-output-names: + maxItems: 1 + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-rx-macro + then: + properties: + clock-names: + oneOf: + - items: # for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: # for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sm6115-lpass-rx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: npl + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sc8280xp-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-rx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: fsgen + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3200000 { + compatible = "qcom,sm8250-lpass-rx-macro"; + reg = <0x3200000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 0>, + <&audiocc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml new file mode 100644 index 000000000..d8682ff2e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -0,0 +1,163 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-tx-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) TX Macro audio codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-tx-macro + - qcom,sm6115-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + - qcom,sm8450-lpass-tx-macro + - qcom,sm8550-lpass-tx-macro + - qcom,sc8280xp-lpass-tx-macro + - items: + - enum: + - qcom,eliza-lpass-tx-macro + - qcom,kaanapali-lpass-tx-macro + - qcom,sm8650-lpass-tx-macro + - qcom,sm8750-lpass-tx-macro + - qcom,x1e80100-lpass-tx-macro + - const: qcom,sm8550-lpass-tx-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 3 + maxItems: 5 + + clock-names: + minItems: 3 + maxItems: 5 + + clock-output-names: + maxItems: 1 + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-tx-macro + then: + properties: + clock-names: + oneOf: + - items: # for ADSP based platforms + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + - items: # for ADSP bypass based platforms + - const: mclk + - const: npl + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sc8280xp-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + - qcom,sm8450-lpass-tx-macro + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sm6115-lpass-tx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: npl + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-tx-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: fsgen + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3220000 { + compatible = "qcom,sm8250-lpass-tx-macro"; + reg = <0x3220000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&aoncc 1>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + qcom,dmic-sample-rate = <600000>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml new file mode 100644 index 000000000..aea31fbda --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -0,0 +1,188 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-va-macro + - qcom,sm6115-lpass-va-macro + - qcom,sm8250-lpass-va-macro + - qcom,sm8450-lpass-va-macro + - qcom,sm8550-lpass-va-macro + - qcom,sc8280xp-lpass-va-macro + - items: + - enum: + - qcom,eliza-lpass-va-macro + - qcom,glymur-lpass-va-macro + - qcom,kaanapali-lpass-va-macro + - qcom,sm8650-lpass-va-macro + - qcom,sm8750-lpass-va-macro + - qcom,x1e80100-lpass-va-macro + - const: qcom,sm8550-lpass-va-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 1 + maxItems: 4 + + clock-names: + minItems: 1 + maxItems: 4 + + clock-output-names: + maxItems: 1 + + power-domains: + maxItems: 2 + + power-domain-names: + items: + - const: macro + - const: dcodec + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + + vdd-micb-supply: + description: phandle to voltage regulator of MIC Bias + +required: + - compatible + - reg + - "#sound-dai-cells" + - clock-names + - clocks + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + const: qcom,sc7280-lpass-va-macro + + then: + if: + required: + - power-domains + then: + properties: + clocks: + maxItems: 1 + clock-names: + items: + - const: mclk + else: + properties: + clocks: + minItems: 3 + maxItems: 3 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - if: + properties: + compatible: + contains: + const: qcom,sm6115-lpass-va-macro + then: + properties: + clocks: + minItems: 3 + maxItems: 3 + clock-names: + items: + - const: mclk + - const: dcodec + - const: npl + - if: + properties: + compatible: + contains: + const: qcom,sm8250-lpass-va-macro + then: + properties: + clocks: + minItems: 3 + maxItems: 3 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + + - if: + properties: + compatible: + contains: + enum: + - qcom,sc8280xp-lpass-va-macro + - qcom,sm8450-lpass-va-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: npl + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-va-macro + then: + properties: + clocks: + minItems: 3 + maxItems: 3 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3370000 { + compatible = "qcom,sm8250-lpass-va-macro"; + reg = <0x3370000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&aoncc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; + clock-names = "mclk", "macro", "dcodec"; + clock-output-names = "fsgen"; + qcom,dmic-sample-rate = <600000>; + vdd-micb-supply = <&vreg_s4a_1p8>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml new file mode 100644 index 000000000..9fedd8053 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.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/qcom,lpass-wsa-macro.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: LPASS(Low Power Audio Subsystem) VA Macro audio codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + compatible: + oneOf: + - enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sm8550-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + - items: + - enum: + - qcom,eliza-lpass-wsa-macro + - qcom,glymur-lpass-wsa-macro + - qcom,kaanapali-lpass-wsa-macro + - qcom,sm8650-lpass-wsa-macro + - qcom,sm8750-lpass-wsa-macro + - qcom,x1e80100-lpass-wsa-macro + - const: qcom,sm8550-lpass-wsa-macro + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + '#clock-cells': + const: 0 + + clocks: + minItems: 4 + maxItems: 6 + + clock-names: + minItems: 4 + maxItems: 6 + + clock-output-names: + maxItems: 1 + + qcom,dmic-sample-rate: + description: dmic sample rate + $ref: /schemas/types.yaml#/definitions/uint32 + + vdd-micb-supply: + description: phandle to voltage regulator of MIC Bias + +required: + - compatible + - reg + - "#sound-dai-cells" + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + contains: + enum: + - qcom,sm8550-lpass-wsa-macro + then: + properties: + clocks: + minItems: 4 + maxItems: 4 + clock-names: + items: + - const: mclk + - const: macro + - const: dcodec + - const: fsgen + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/qcom,sm8250-lpass-aoncc.h> + #include <dt-bindings/sound/qcom,q6afe.h> + codec@3240000 { + compatible = "qcom,sm8250-lpass-wsa-macro"; + reg = <0x3240000 0x1000>; + #sound-dai-cells = <1>; + #clock-cells = <0>; + clocks = <&audiocc 1>, + <&audiocc 0>, + <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&vamacro>; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-output-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital-codec.yaml b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital-codec.yaml new file mode 100644 index 000000000..33bc23b61 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital-codec.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,msm8916-wcd-digital-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm MSM8916 WCD Digital Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + The digital WCD audio codec found on Qualcomm MSM8916 LPASS. + +properties: + compatible: + oneOf: + - items: + - enum: + - qcom,sdm660-wcd-digital-codec + - const: qcom,msm8916-wcd-digital-codec + + - const: qcom,msm8916-wcd-digital-codec + + reg: + maxItems: 1 + + clocks: + maxItems: 2 + + clock-names: + items: + - const: ahbix-clk + - const: mclk + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - clocks + - clock-names + - '#sound-dai-cells' + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/qcom,gcc-msm8916.h> + audio-codec@771c000 { + compatible = "qcom,msm8916-wcd-digital-codec"; + reg = <0x0771c000 0x400>; + clocks = <&gcc GCC_ULTAUDIO_AHBFABRIC_IXFABRIC_CLK>, + <&gcc GCC_CODEC_DIGCODEC_CLK>; + clock-names = "ahbix-clk", "mclk"; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,pm4125-codec.yaml b/Documentation/devicetree/bindings/sound/qcom,pm4125-codec.yaml new file mode 100644 index 000000000..6e2f103be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,pm4125-codec.yaml @@ -0,0 +1,134 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,pm4125-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm PM4125 Audio Codec + +maintainers: + - Alexey Klimov <alexey.klimov@linaro.org> + +description: + The audio codec IC found on Qualcomm PM4125/PM2250 PMIC. + It has RX and TX Soundwire slave devices. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: qcom,pm4125-codec + + reg: + description: + Specifies the SPMI base address for the audio codec peripherals. The + address space contains reset register needed to power-on the codec. + maxItems: 1 + + reg-names: + maxItems: 1 + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-cp-supply: + description: A reference to the charge pump I/O supply + + vdd-mic-bias-supply: + description: A reference to the 3.3V mic bias supply + + vdd-pa-vpos-supply: + description: A reference to the PA VPOS 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,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 + - reg + - vdd-io-supply + - vdd-cp-supply + - vdd-mic-bias-supply + - vdd-pa-vpos-supply + - qcom,tx-device + - qcom,rx-device + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/spmi/spmi.h> + + spmi { + #address-cells = <2>; + #size-cells = <0>; + + pmic { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@f000 { + compatible = "qcom,pm4125-codec"; + reg = <0xf000>; + vdd-io-supply = <&pm4125_l15>; + vdd-cp-supply = <&pm4125_s4>; + vdd-pa-vpos-supply = <&pm4125_s4>; + vdd-mic-bias-supply = <&pm4125_l22>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,rx-device = <&pm4125_rx>; + qcom,tx-device = <&pm4125_tx>; + #sound-dai-cells = <1>; + }; + }; + }; + + /* ... */ + + soundwire@a610000 { + reg = <0x0a610000 0x2000>; + #address-cells = <2>; + #size-cells = <0>; + pm4125_rx: audio-codec@0,4 { + compatible = "sdw20217010c00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 3>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.yaml new file mode 100644 index 000000000..769e4cb5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.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/qcom,pm4125-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm SoundWire Slave devices on PM4125/PM2250 PMIC audio codec. + +maintainers: + - Alexey Klimov <alexey.klimov@linaro.org> + +description: + The audio codec IC found on Qualcomm PM4125/PM2250 PMICs. + It has RX and TX Soundwire slave devices. + +properties: + compatible: + const: sdw20217010c00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between device and host tx ports. + In the order of the device port index which are adc1_port, adc23_port, + dmic03_mbhc_port, dmic46_port. + Supports maximum 2 tx soundwire ports. + + PM4125 TX Port 1 (ADC1,2 & DMIC0 & MBHC) <=> SWR0 Port 1 + PM4125 TX Port 2 (ADC1 & DMIC0,1,2 & MBHC) <=> SWR0 Port 2 + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 4 + items: + enum: [1, 2, 3, 4] + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between device and host rx ports. + In the order of device port index which are hph_port, clsh_port, + comp_port, lo_port, dsd port. + Supports maximum 2 rx soundwire ports. + + PM4125 RX Port 1 (HPH_L/R) <==> SWR1 Port 1 (HPH_L/R) + PM4125 RX Port 2 (COMP_L/R) <==> SWR1 Port 3 (COMP_L/R) + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 5 + items: + enum: [1, 2, 3, 4, 5] + +required: + - compatible + - reg + +oneOf: + - required: + - qcom,tx-port-mapping + - required: + - qcom,rx-port-mapping + +additionalProperties: false + +examples: + - | + soundwire@a610000 { + reg = <0x0a610000 0x2000>; + #address-cells = <2>; + #size-cells = <0>; + pm4125_rx: codec@0,1 { + compatible = "sdw20217010c00"; + reg = <0 1>; + qcom,rx-port-mapping = <1 3>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml b/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml new file mode 100644 index 000000000..be47dbdb2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml @@ -0,0 +1,162 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + oneOf: + - items: + - enum: + - qcom,pm660l-wcd-analog-codec + - const: qcom,pm8953-wcd-analog-codec + + - enum: + - qcom,pm8916-wcd-analog-codec + - qcom,pm8950-wcd-analog-codec + - qcom,pm8953-wcd-analog-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 14 + + interrupt-names: + items: + - const: cdc_spk_cnp_int + - const: cdc_spk_clip_int + - const: cdc_spk_ocp_int + - const: mbhc_ins_rem_det1 + - const: mbhc_but_rel_det + - const: mbhc_but_press_det + - const: mbhc_ins_rem_det + - const: mbhc_switch_int + - const: cdc_ear_ocp_int + - const: cdc_hphr_ocp_int + - const: cdc_hphl_ocp_det + - const: cdc_ear_cnp_int + - const: cdc_hphr_cnp_int + - const: cdc_hphl_cnp_int + + vdd-cdc-io-supply: + description: 1.8V buck supply + + vdd-cdc-tx-rx-cx-supply: + description: 1.8V SIDO buck supply + + vdd-micbias-supply: + description: micbias supply + + qcom,mbhc-vthreshold-low: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Array of 5 threshold voltages in mV for 5-button detection on + headset when MBHC is powered by an internal current source. + minItems: 5 + maxItems: 5 + + qcom,mbhc-vthreshold-high: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Array of 5 threshold voltages in mV for 5-button detection on + headset when MBHC is powered from micbias. + minItems: 5 + maxItems: 5 + + qcom,micbias-lvl: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Voltage (mV) for Mic Bias + + qcom,hphl-jack-type-normally-open: + type: boolean + description: + True if the HPHL pin on the jack is NO (Normally Open), false if it's + NC (Normally Closed). + + qcom,gnd-jack-type-normally-open: + type: boolean + description: + True if the GND pin on the jack is NO (Normally Open), false if it's + NC (Normally Closed). + + qcom,micbias1-ext-cap: + type: boolean + description: + True if micbias1 has an external capacitor. + + qcom,micbias2-ext-cap: + type: boolean + description: + True if micbias2 has an external capacitor. + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/spmi/spmi.h> + + pmic@1 { + compatible = "qcom,pm8916", "qcom,spmi-pmic"; + reg = <0x1 SPMI_USID>; + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@f000 { + compatible = "qcom,pm8916-wcd-analog-codec"; + reg = <0xf000>; + qcom,mbhc-vthreshold-low = <75 150 237 450 500>; + qcom,mbhc-vthreshold-high = <75 150 237 450 500>; + interrupt-parent = <&spmi_bus>; + interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, + <0x1 0xf0 0x1 IRQ_TYPE_NONE>, + <0x1 0xf0 0x2 IRQ_TYPE_NONE>, + <0x1 0xf0 0x3 IRQ_TYPE_NONE>, + <0x1 0xf0 0x4 IRQ_TYPE_NONE>, + <0x1 0xf0 0x5 IRQ_TYPE_NONE>, + <0x1 0xf0 0x6 IRQ_TYPE_NONE>, + <0x1 0xf0 0x7 IRQ_TYPE_NONE>, + <0x1 0xf1 0x0 IRQ_TYPE_NONE>, + <0x1 0xf1 0x1 IRQ_TYPE_NONE>, + <0x1 0xf1 0x2 IRQ_TYPE_NONE>, + <0x1 0xf1 0x3 IRQ_TYPE_NONE>, + <0x1 0xf1 0x4 IRQ_TYPE_NONE>, + <0x1 0xf1 0x5 IRQ_TYPE_NONE>; + interrupt-names = "cdc_spk_cnp_int", + "cdc_spk_clip_int", + "cdc_spk_ocp_int", + "mbhc_ins_rem_det1", + "mbhc_but_rel_det", + "mbhc_but_press_det", + "mbhc_ins_rem_det", + "mbhc_switch_int", + "cdc_ear_ocp_int", + "cdc_hphr_ocp_int", + "cdc_hphl_ocp_det", + "cdc_ear_cnp_int", + "cdc_hphr_cnp_int", + "cdc_hphl_cnp_int"; + vdd-cdc-io-supply = <&pm8916_l5>; + vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>; + vdd-micbias-supply = <&pm8916_l13>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml b/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml new file mode 100644 index 000000000..26fe8cc66 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml @@ -0,0 +1,39 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6adm-routing.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Device Manager (Q6ADM) routing + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Qualcomm Audio Device Manager (Q6ADM) routing node represents routing + specific configuration. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - qcom,q6adm-routing + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + routing { + compatible = "qcom,q6adm-routing"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6adm.yaml new file mode 100644 index 000000000..3c32c5b0f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6adm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Device Manager (Q6ADM) + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6adm + + routing: + type: object + $ref: /schemas/sound/qcom,q6adm-routing.yaml# + unevaluatedProperties: false + description: Qualcomm DSP LPASS audio routing + +required: + - compatible + - routing + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@8 { + compatible = "qcom,q6adm"; + reg = <APR_SVC_ADM>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + routing { + compatible = "qcom,q6adm-routing"; + #sound-dai-cells = <0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.yaml b/Documentation/devicetree/bindings/sound/qcom,q6afe.yaml new file mode 100644 index 000000000..4624b3d46 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$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 <krzk@kernel.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 + + usbd: + type: object + $ref: /schemas/sound/qcom,q6usb.yaml# + unevaluatedProperties: false + description: Qualcomm DSP USB audio ports + +required: + - compatible + - dais + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + #include <dt-bindings/sound/qcom,q6afe.h> + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + clock-controller { + compatible = "qcom,q6afe-clocks"; + #clock-cells = <2>; + }; + + dais { + compatible = "qcom,q6afe-dais"; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@22 { + reg = <QUATERNARY_MI2S_RX>; + qcom,sd-lines = <0 1 2 3>; + }; + }; + + usbd { + compatible = "qcom,q6usb"; + #sound-dai-cells = <1>; + iommus = <&apps_smmu 0x180f 0x0>; + qcom,usb-audio-intr-idx = /bits/ 16 <2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml new file mode 100644 index 000000000..9e5b30d9c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml @@ -0,0 +1,34 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6apm-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Process Manager Digital Audio Interfaces + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm APM DAIs in DSP + +properties: + compatible: + const: qcom,q6apm-dais + + iommus: + minItems: 1 + maxItems: 2 + +required: + - compatible + - iommus + +additionalProperties: false + +examples: + - | + dais { + compatible = "qcom,q6apm-dais"; + iommus = <&apps_smmu 0x1801 0x0>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml new file mode 100644 index 000000000..4878c424e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$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 <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - qcom,q6apm-lpass-dais + + '#sound-dai-cells': + const: 1 + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + +# Digital Audio Interfaces +patternProperties: + '^dai@[0-9a-f]+$': + type: object + description: + Q6DSP Digital Audio Interfaces. + + properties: + reg: + maxItems: 1 + description: + Digital Audio Interface ID + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + minItems: 1 + items: + - enum: [bclk, mclk] + - const: mclk + + dependencies: + clocks: [clock-names] + + required: + - reg + + additionalProperties: false + +required: + - compatible + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6afe.h> + + dais { + compatible = "qcom,q6apm-lpass-dais"; + #sound-dai-cells = <1>; + #address-cells = <1>; + #size-cells = <0>; + + dai@10 { + reg = <PRIMARY_MI2S_RX>; + clocks = <&q6prmcc LPASS_CLK_ID_PRI_MI2S_IBIT + LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&q6prmcc LPASS_CLK_ID_MCLK_1 + LPASS_CLK_ATTRIBUTE_COUPLE_NO>; + clock-names = "bclk", "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm.yaml new file mode 100644 index 000000000..ec06769a2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6apm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Process Manager (Q6APM) + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6apm + + bedais: + type: object + $ref: /schemas/sound/qcom,q6apm-lpass-dais.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + + dais: + type: object + $ref: /schemas/sound/qcom,q6apm-dai.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - bedais + - dais + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,gpr.h> + + gpr { + #address-cells = <1>; + #size-cells = <0>; + + service@1 { + reg = <GPR_APM_MODULE_IID>; + compatible = "qcom,q6apm"; + #sound-dai-cells = <0>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + dais { + compatible = "qcom,q6apm-dais"; + iommus = <&apps_smmu 0x1801 0x0>; + }; + + bedais { + compatible = "qcom,q6apm-lpass-dais"; + #sound-dai-cells = <1>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml b/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml new file mode 100644 index 000000000..bc8c8ba24 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml @@ -0,0 +1,96 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6asm-dais.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Stream Manager (Q6ASM) + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Q6ASM is one of the APR audio services on Q6DSP. Each of its subnodes + represent a dai with board specific configuration. + +properties: + compatible: + enum: + - qcom,q6asm-dais + + iommus: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + +patternProperties: + "^dai@[0-9]+$": + type: object + description: + Q6ASM Digital Audio Interface + + properties: + reg: + maxItems: 1 + + direction: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: | + The direction of the dai stream: + - Q6ASM_DAI_TX_RX (0) for both tx and rx + - Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + - Q6ASM_DAI_RX (2) for only rx (Playback/Decode) + + is-compress-dai: + type: boolean + description: + Compress offload dai. + + dependencies: + is-compress-dai: [ direction ] + + required: + - reg + + additionalProperties: false + +required: + - compatible + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + dais { + compatible = "qcom,q6asm-dais"; + iommus = <&apps_smmu 0x1821 0x0>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@0 { + reg = <0>; + }; + + dai@1 { + reg = <1>; + }; + + dai@2 { + reg = <2>; + is-compress-dai; + direction = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6asm.yaml new file mode 100644 index 000000000..a6f88ce92 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6asm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Stream Manager (Q6ASM) + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6asm + + dais: + type: object + $ref: /schemas/sound/qcom,q6asm-dais.yaml# + unevaluatedProperties: false + description: Qualcomm DSP audio ports + +required: + - compatible + - dais + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@7 { + compatible = "qcom,q6asm"; + reg = <APR_SVC_ASM>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + dais { + compatible = "qcom,q6asm-dais"; + iommus = <&apps_smmu 0x1821 0x0>; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@0 { + reg = <0>; + }; + + dai@1 { + reg = <1>; + }; + + dai@2 { + reg = <2>; + is-compress-dai; + direction = <1>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.yaml b/Documentation/devicetree/bindings/sound/qcom,q6core.yaml new file mode 100644 index 000000000..8642ef9f9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6core.yaml @@ -0,0 +1,39 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6core.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Audio Core (Q6Core) + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6core + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,apr.h> + + apr { + #address-cells = <1>; + #size-cells = <0>; + + service@3 { + compatible = "qcom,q6core"; + reg = <APR_SVC_ADSP_CORE>; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml new file mode 100644 index 000000000..3552c4413 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-clocks.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm DSP LPASS Clock Controller + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + This binding describes the Qualcomm DSP Clock Controller + +properties: + compatible: + enum: + - qcom,q6afe-clocks + - qcom,q6prm-lpass-clocks + + '#clock-cells': + const: 2 + description: + Clock Id is followed by clock coupling attributes. + 1 = for no coupled clock + 2 = for dividend of the coupled clock + 3 = for divisor of the coupled clock + 4 = for inverted and no couple clock + +required: + - compatible + - "#clock-cells" + +additionalProperties: false + +examples: + - | + clock-controller { + compatible = "qcom,q6afe-clocks"; + #clock-cells = <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml new file mode 100644 index 000000000..3b03e2acd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml @@ -0,0 +1,167 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + Indicate how data is packed within the slot. For example, 32 slot + width in case of sample bit width is 24TDM Invert the sync. + 0 = MSB + 1 = LSB + + required: + - reg + + allOf: + - if: + properties: + reg: + contains: + # TDM DAI ID range from PRIMARY_TDM_RX_0 - QUINARY_TDM_TX_7 + items: + minimum: 24 + maximum: 103 + then: + required: + - qcom,tdm-sync-mode + - qcom,tdm-sync-src + - qcom,tdm-data-out + - qcom,tdm-invert-sync + - qcom,tdm-data-delay + - qcom,tdm-data-align + + - if: + properties: + reg: + contains: + # MI2S DAI ID range PRIMARY_MI2S_RX - QUATERNARY_MI2S_TX and + # QUINARY_MI2S_RX - QUINARY_MI2S_TX and + # LPI_MI2S_RX_0 - LPI_MI2S_TX_6 + items: + oneOf: + - minimum: 16 + maximum: 23 + - minimum: 127 + maximum: 128 + - minimum: 137 + maximum: 152 + then: + required: + - qcom,sd-lines + + additionalProperties: false + +required: + - compatible + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,q6dsp-lpass-ports.h> + + dais { + compatible = "qcom,q6afe-dais"; + #address-cells = <1>; + #size-cells = <0>; + #sound-dai-cells = <1>; + + dai@22 { + reg = <QUATERNARY_MI2S_RX>; + qcom,sd-lines = <0 1 2 3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6prm.yaml b/Documentation/devicetree/bindings/sound/qcom,q6prm.yaml new file mode 100644 index 000000000..3eafe189e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6prm.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6prm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Proxy Resource Manager (Q6PRM) + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +allOf: + - $ref: /schemas/soc/qcom/qcom,apr-services.yaml# + +properties: + compatible: + enum: + - qcom,q6prm + + clock-controller: + $ref: /schemas/sound/qcom,q6dsp-lpass-clocks.yaml# + unevaluatedProperties: false + description: Qualcomm DSP LPASS clock controller + +required: + - compatible + - clock-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/soc/qcom,gpr.h> + + gpr { + #address-cells = <1>; + #size-cells = <0>; + + service@2 { + reg = <GPR_PRM_MODULE_IID>; + compatible = "qcom,q6prm"; + qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd"; + + clock-controller { + compatible = "qcom,q6prm-lpass-clocks"; + #clock-cells = <2>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6usb.yaml b/Documentation/devicetree/bindings/sound/qcom,q6usb.yaml new file mode 100644 index 000000000..37161d2aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6usb.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,q6usb.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm ASoC DPCM USB backend DAI + +maintainers: + - Wesley Cheng <quic_wcheng@quicinc.com> + +description: + The USB port is a supported AFE path on the Q6 DSP. This ASoC DPCM + backend DAI will communicate the required settings to initialize the + XHCI host controller properly for enabling the offloaded audio stream. + Parameters defined under this node will carry settings, which will be + passed along during the QMI stream enable request and configuration of + the XHCI host controller. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - qcom,q6usb + + iommus: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + + qcom,usb-audio-intr-idx: + description: + Desired XHCI interrupter number to use. Depending on the audio DSP + on the platform, it will operate on a specific XHCI interrupter. + $ref: /schemas/types.yaml#/definitions/uint16 + maximum: 8 + +required: + - compatible + - "#sound-dai-cells" + - qcom,usb-audio-intr-idx + +additionalProperties: false + +examples: + - | + dais { + compatible = "qcom,q6usb"; + #sound-dai-cells = <1>; + iommus = <&apps_smmu 0x180f 0x0>; + qcom,usb-audio-intr-idx = /bits/ 16 <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml new file mode 100644 index 000000000..1536fcd96 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -0,0 +1,249 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,sm8250.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. ASoC sound card drivers + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + This bindings describes Qualcomm SoC based sound cards + which uses LPASS internal codec for audio. + +properties: + compatible: + oneOf: + - items: + - enum: + - lenovo,yoga-c630-sndcard + - qcom,db845c-sndcard + - const: qcom,sdm845-sndcard + - items: + - enum: + - qcom,eliza-sndcard + - qcom,hawi-sndcard + - qcom,kaanapali-sndcard + - qcom,maili-sndcard + - qcom,sm8475-sndcard + - qcom,sm8550-sndcard + - qcom,sm8650-sndcard + - qcom,sm8750-sndcard + - const: qcom,sm8450-sndcard + - enum: + - ayaneo,pocket-s2-sndcard + - fairphone,fp4-sndcard + - fairphone,fp5-sndcard + - qcom,apq8096-sndcard + - qcom,glymur-sndcard + - qcom,qcm6490-idp-sndcard + - qcom,qcs615-sndcard + - qcom,qcs6490-rb3gen2-sndcard + - qcom,qcs8275-sndcard + - qcom,qcs9075-sndcard + - qcom,qcs9100-sndcard + - qcom,qrb2210-sndcard + - qcom,qrb4210-rb2-sndcard + - qcom,qrb5165-rb5-sndcard + - qcom,sc7180-qdsp6-sndcard + - qcom,sc8280xp-sndcard + - qcom,sdm660-sndcard + - qcom,sdm845-sndcard + - qcom,sm8250-sndcard + - qcom,sm8450-sndcard + - qcom,x1e80100-sndcard + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + +patternProperties: + ".*-dai-link$": + 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 + + dai-tdm-slot-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Number of slots in use + + dai-tdm-slot-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Width, in bits, of each slot + + patternProperties: + '^dai-tdm-slot-[rt]x-mask$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Slot mask for active TDM slots + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + dai-tdm-slot-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Number of slots in use + + dai-tdm-slot-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Width, in bits, of each slot + + patternProperties: + '^dai-tdm-slot-[rt]x-mask$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Slot mask for active TDM slots + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - model + +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>; + }; + }; + + usb-dai-link { + link-name = "USB Playback"; + cpu { + sound-dai = <&q6afedai USB_RX>; + }; + + codec { + sound-dai = <&usbdai USB_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd9335.yaml new file mode 100644 index 000000000..34f8fe4da --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.yaml @@ -0,0 +1,156 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd9335.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WCD9335 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC with in-built + Soundwire controller and interrupt mux. It supports both I2S/I2C and SLIMbus + audio interfaces. + +properties: + compatible: + const: slim217,1a0 + + reg: + maxItems: 1 + + clocks: + maxItems: 2 + + clock-names: + items: + - const: mclk + - const: slimbus + + interrupts: + maxItems: 2 + + interrupt-names: + items: + - const: intr1 + - const: intr2 + + interrupt-controller: true + + '#interrupt-cells': + const: 1 + + reset-gpios: + maxItems: 1 + + slim-ifc-dev: + description: SLIM IFC device interface + $ref: /schemas/types.yaml#/definitions/phandle + + '#sound-dai-cells': + const: 1 + + vdd-buck-supply: + description: 1.8V buck supply + + vdd-buck-sido-supply: + description: 1.8V SIDO buck supply + + vdd-io-supply: + description: 1.8V I/O supply + + vdd-micbias-supply: + description: micbias supply + + vdd-rx-supply: + description: 1.8V rx supply + + vdd-tx-supply: + description: 1.8V tx supply + + vdd-vbat-supply: + description: vbat supply + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + required: + - slim-ifc-dev + then: + required: + - clocks + - clock-names + - interrupts + - interrupt-names + - interrupt-controller + - '#interrupt-cells' + - reset-gpios + - slim-ifc-dev + - '#sound-dai-cells' + - vdd-buck-supply + - vdd-buck-sido-supply + - vdd-io-supply + - vdd-rx-supply + - vdd-tx-supply + else: + properties: + clocks: false + clock-names: false + interrupts: false + interrupt-names: false + interrupt-controller: false + '#interrupt-cells': false + reset-gpios: false + slim-ifc-dev: false + '#sound-dai-cells': false + vdd-buck-supply: false + vdd-buck-sido-supply: false + vdd-io-supply: false + vdd-micbias-supply: false + vdd-rx-supply: false + vdd-tx-supply: false + vdd-vbat-supply: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/qcom,rpmcc.h> + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + tasha_ifd: codec@0,0 { + compatible = "slim217,1a0"; + reg = <0 0>; + }; + + codec@1,0 { + compatible = "slim217,1a0"; + reg = <1 0>; + + clock-names = "mclk", "slimbus"; + clocks = <&div1_mclk>, <&rpmcc RPM_SMD_BB_CLK1>; + + interrupt-parent = <&tlmm>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH>, + <53 IRQ_TYPE_LEVEL_HIGH>; + interrupt-names = "intr1", "intr2"; + interrupt-controller; + #interrupt-cells = <1>; + + reset-gpios = <&tlmm 64 GPIO_ACTIVE_LOW>; + slim-ifc-dev = <&tasha_ifd>; + #sound-dai-cells = <1>; + + vdd-buck-supply = <&vreg_s4a_1p8>; + vdd-buck-sido-supply = <&vreg_s4a_1p8>; + vdd-tx-supply = <&vreg_s4a_1p8>; + vdd-rx-supply = <&vreg_s4a_1p8>; + vdd-io-supply = <&vreg_s4a_1p8>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml new file mode 100644 index 000000000..3a7334e41 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -0,0 +1,239 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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: + - | + #include <dt-bindings/gpio/gpio.h> + codec@1,0{ + compatible = "slim217,250"; + reg = <1 0>; + reset-gpios = <&tlmm 64 GPIO_ACTIVE_LOW>; + slim-ifc-dev = <&wcd9340_ifd>; + #sound-dai-cells = <1>; + interrupt-parent = <&tlmm>; + interrupts = <54 4>; + interrupt-controller; + #interrupt-cells = <1>; + #clock-cells = <0>; + clock-frequency = <9600000>; + clock-output-names = "mclk"; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headset-vthreshold-microvolt = <1700000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + clock-names = "extclk"; + clocks = <&rpmhcc 2>; + + #address-cells = <1>; + #size-cells = <1>; + + gpio@42 { + compatible = "qcom,wcd9340-gpio"; + reg = <0x42 0x2>; + gpio-controller; + #gpio-cells = <2>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml new file mode 100644 index 000000000..c8543f969 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd937x-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm SoundWire Slave devices on WCD9370/WCD9375 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9370/WCD9375 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: sdw20217010a00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between device and host tx ports. + In the order of the device port index which are adc1_port, adc23_port, + dmic03_mbhc_port, dmic46_port. + Supports maximum 4 tx soundwire ports. + + WCD9370 TX Port 1 (ADC1) <=> SWR2 Port 2 + WCD9370 TX Port 2 (ADC2, 3) <=> SWR2 Port 2 + WCD9370 TX Port 3 (DMIC0,1,2,3 & MBHC) <=> SWR2 Port 3 + WCD9370 TX Port 4 (DMIC4,5,6,7) <=> SWR2 Port 4 + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + items: + enum: [1, 2, 3, 4] + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between device and host rx ports. + In the order of device port index which are hph_port, clsh_port, + comp_port, lo_port, dsd port. + Supports maximum 5 rx soundwire ports. + + WCD9370 RX Port 1 (HPH_L/R) <==> SWR1 Port 1 (HPH_L/R) + WCD9370 RX Port 2 (CLSH) <==> SWR1 Port 2 (CLSH) + WCD9370 RX Port 3 (COMP_L/R) <==> SWR1 Port 3 (COMP_L/R) + WCD9370 RX Port 4 (LO) <==> SWR1 Port 4 (LO) + WCD9370 RX Port 5 (DSD_L/R) <==> SWR1 Port 5 (DSD) + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + items: + enum: [1, 2, 3, 4, 5] + + qcom,tx-channel-mapping: + description: | + Specifies static channel mapping between slave and master tx port + channels. + In the order of slave port channels which is adc1, adc2, adc3, + dmic0, dmic1, mbhc, dmic2, dmic3, dmci4, dmic5, dmic6, dmic7. + $ref: /schemas/types.yaml#/definitions/uint8-array + minItems: 12 + maxItems: 12 + additionalItems: false + items: + enum: + - 1 # WCD9370_SWRM_CH1 + - 2 # WCD9370_SWRM_CH2 + - 3 # WCD9370_SWRM_CH3 + - 4 # WCD9370_SWRM_CH4 + + qcom,rx-channel-mapping: + description: | + Specifies static channels mapping between slave and master rx port + channels. + In the order of slave port channels, which is + hph_l, hph_r, clsh, comp_l, comp_r, lo, dsd_r, dsd_l. + $ref: /schemas/types.yaml#/definitions/uint8-array + minItems: 8 + maxItems: 8 + additionalItems: false + items: + enum: + - 1 # WCD9370_SWRM_CH1 + - 2 # WCD9370_SWRM_CH2 + - 3 # WCD9370_SWRM_CH3 + - 4 # WCD9370_SWRM_CH4 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + reg = <0x03210000 0x2000>; + #address-cells = <2>; + #size-cells = <0>; + wcd937x_rx: codec@0,4 { + compatible = "sdw20217010a00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + qcom,rx-channel-mapping = /bits/ 8 <1 2 1 1 2 1 1 2>; + }; + }; + + soundwire@3230000 { + reg = <0x03230000 0x2000>; + #address-cells = <2>; + #size-cells = <0>; + wcd937x_tx: codec@0,3 { + compatible = "sdw20217010a00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 2 3 4>; + qcom,tx-channel-mapping = /bits/ 8 <1 2 1 1 2 3 3 4 1 2 3 4>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml new file mode 100644 index 000000000..f94203798 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd937x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WCD9370/WCD9375 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. + +allOf: + - $ref: dai-common.yaml# + - $ref: qcom,wcd93xx-common.yaml# + +properties: + compatible: + oneOf: + - const: qcom,wcd9370-codec + - items: + - const: qcom,wcd9375-codec + - const: qcom,wcd9370-codec + + vdd-px-supply: + description: A reference to the 1.8V I/O supply + +required: + - compatible + - vdd-px-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "qcom,wcd9370-codec"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&wcd_reset_n>; + pinctrl-1 = <&wcd_reset_n_sleep>; + reset-gpios = <&tlmm 83 GPIO_ACTIVE_LOW>; + vdd-buck-supply = <&vreg_l17b_1p8>; + vdd-rxtx-supply = <&vreg_l18b_1p8>; + vdd-px-supply = <&vreg_l18b_1p8>; + vdd-mic-bias-supply = <&vreg_bob>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,rx-device = <&wcd937x_rx>; + qcom,tx-device = <&wcd937x_tx>; + #sound-dai-cells = <1>; + }; + + /* ... */ + + soundwire@3210000 { + reg = <0x03210000 0x2000>; + #address-cells = <2>; + #size-cells = <0>; + wcd937x_rx: codec@0,4 { + compatible = "sdw20217010a00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + reg = <0x03230000 0x2000>; + #address-cells = <2>; + #size-cells = <0>; + wcd937x_tx: codec@0,3 { + compatible = "sdw20217010a00"; + reg = <0 3>; + qcom,tx-port-mapping = <1 2 3 4>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml new file mode 100644 index 000000000..7b31bf93f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm SoundWire Slave devices on WCD9380/WCD9385 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. This bindings is for the + slave devices. + +properties: + compatible: + const: sdw20217010d00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between slave and master tx ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between slave and master rx ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml new file mode 100644 index 000000000..ab1c6285d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd938x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WCD9380/WCD9385 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire slave devices. + +allOf: + - $ref: dai-common.yaml# + - $ref: qcom,wcd93xx-common.yaml# + +properties: + compatible: + enum: + - qcom,wcd9380-codec + - qcom,wcd9385-codec + + mux-controls: + description: A reference to the audio mux switch for + switching CTIA/OMTP Headset types + maxItems: 1 + + us-euro-gpios: + description: GPIO spec for swapping gnd and mic segments + maxItems: 1 + deprecated: true + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "qcom,wcd9380-codec"; + reset-gpios = <&tlmm 32 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <1>; + qcom,tx-device = <&wcd938x_tx>; + qcom,rx-device = <&wcd938x_rx>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + }; + + /* ... */ + + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010d00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010d00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml new file mode 100644 index 000000000..67ed7701b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd939x-sdw.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm SoundWire devices on WCD9390/WCD9395 + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9390/WCD9395 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire devices. This bindings is for the devices. + +properties: + compatible: + const: sdw20217010e00 + + reg: + maxItems: 1 + + qcom,tx-port-mapping: + description: | + Specifies static port mapping between device and host tx ports. + In the order of the device port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + + qcom,rx-port-mapping: + description: | + Specifies static port mapping between device and host rx ports. + In the order of device port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd938x_rx: codec@0,4 { + compatible = "sdw20217010e00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5 6>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010e00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml new file mode 100644 index 000000000..85283f944 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd939x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WCD9380/WCD9385 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: | + Qualcomm WCD9390/WCD9395 Codec is a standalone Hi-Fi audio codec IC. + It has RX and TX Soundwire devices. + The WCD9390/WCD9395 IC has a functionally separate USB-C Mux subsystem + accessible over an I2C interface. + The Audio Headphone and Microphone data path between the Codec and the USB-C Mux + subsystems are external to the IC, thus requiring DT port-endpoint graph description + to handle USB-C altmode & orientation switching for Audio Accessory Mode. + +allOf: + - $ref: dai-common.yaml# + - $ref: qcom,wcd93xx-common.yaml# + +properties: + compatible: + oneOf: + - const: qcom,wcd9390-codec + - items: + - const: qcom,wcd9395-codec + - const: qcom,wcd9390-codec + + mode-switch: + description: Flag the port as possible handler of altmode switching + type: boolean + + orientation-switch: + description: Flag the port as possible handler of orientation switching + type: boolean + + port: + $ref: /schemas/graph.yaml#/properties/port + description: + A port node to link the WCD939x Codec node to USB MUX subsystems for the + purpose of handling altmode muxing and orientation switching to detect and + enable Audio Accessory Mode. + + vdd-px-supply: + description: A reference to the 1.2V PX supply + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "qcom,wcd9390-codec"; + reset-gpios = <&tlmm 32 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <1>; + qcom,tx-device = <&wcd939x_tx>; + qcom,rx-device = <&wcd939x_rx>; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + qcom,hphl-jack-type-normally-closed; + qcom,ground-jack-type-normally-closed; + qcom,mbhc-buttons-vthreshold-microvolt = <75000 150000 237000 500000 500000 500000 500000 500000>; + qcom,mbhc-headphone-vthreshold-microvolt = <50000>; + }; + + /* ... */ + + soundwire@3210000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03210000 0x2000>; + wcd939x_rx: codec@0,4 { + compatible = "sdw20217010e00"; + reg = <0 4>; + qcom,rx-port-mapping = <1 2 3 4 5 6>; + }; + }; + + soundwire@3230000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x03230000 0x2000>; + wcd938x_tx: codec@0,3 { + compatible = "sdw20217010e00"; + reg = <0 3>; + qcom,tx-port-mapping = <2 3 4 5>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml new file mode 100644 index 000000000..f78ba148a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml @@ -0,0 +1,95 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd93xx-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Common properties for Qualcomm WCD93xx Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +properties: + reset-gpios: + description: GPIO spec for reset line to use + maxItems: 1 + + vdd-buck-supply: + description: A reference to the 1.8V buck supply + + vdd-rxtx-supply: + description: A reference to the 1.8V rx supply + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-mic-bias-supply: + description: A reference to the 3.8V mic bias supply + + qcom,tx-device: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A reference to Soundwire tx device phandle + + qcom,rx-device: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A reference to Soundwire rx device phandle + + qcom,micbias1-microvolt: + description: micbias1 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias2-microvolt: + description: micbias2 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias3-microvolt: + description: micbias3 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias4-microvolt: + description: micbias4 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,hphl-jack-type-normally-closed: + description: Indicates that HPHL jack switch type is normally closed + type: boolean + + qcom,ground-jack-type-normally-closed: + description: Indicates that Headset Ground switch type is normally closed + type: boolean + + qcom,mbhc-headset-vthreshold-microvolt: + description: Voltage threshold value for headset detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-headphone-vthreshold-microvolt: + description: Voltage threshold value for headphone detection + minimum: 0 + maximum: 2850000 + + qcom,mbhc-buttons-vthreshold-microvolt: + description: + Array of 8 Voltage threshold values corresponding to headset + button0 - button7 + minItems: 8 + maxItems: 8 + + '#sound-dai-cells': + const: 1 + +required: + - reset-gpios + - qcom,tx-device + - qcom,rx-device + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#sound-dai-cells" + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml new file mode 100644 index 000000000..ac03672eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw10217201000 + + reg: + maxItems: 1 + + powerdown-gpios: + description: GPIO spec for Powerdown/Shutdown line to use + maxItems: 1 + + '#thermal-sensor-cells': + const: 0 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - powerdown-gpios + - "#thermal-sensor-cells" + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + soundwire@c2d0000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x0c2d0000 0x2000>; + + speaker@0,1 { + compatible = "sdw10217201000"; + reg = <0 1>; + powerdown-gpios = <&wcdpinctrl 2 0>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + + speaker@0,2 { + compatible = "sdw10217201000"; + reg = <0 2>; + powerdown-gpios = <&wcdpinctrl 2 0>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml new file mode 100644 index 000000000..098f1df62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + reset-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + vdd-supply: + description: VDD Supply for the Codec + + qcom,port-mapping: + description: | + Specifies static port mapping between slave and master ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + + '#thermal-sensor-cells': + const: 0 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - vdd-supply + - "#thermal-sensor-cells" + - "#sound-dai-cells" + +oneOf: + - required: + - powerdown-gpios + - required: + - reset-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire@3250000 { + #address-cells = <2>; + #size-cells = <0>; + reg = <0x3250000 0x2000>; + + speaker@0,1 { + compatible = "sdw10217020200"; + reg = <0 1>; + powerdown-gpios = <&tlmm 1 GPIO_ACTIVE_LOW>; + vdd-supply = <&vreg_s10b_1p8>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrLeft"; + }; + + speaker@0,2 { + compatible = "sdw10217020200"; + reg = <0 2>; + powerdown-gpios = <&tlmm 89 GPIO_ACTIVE_LOW>; + vdd-supply = <&vreg_s10b_1p8>; + #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000..866c5e780 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Srinivas Kandagatla <srinivas.kandagatla@linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + reset-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + qcom,port-mapping: + description: | + Specifies static port mapping between slave and master ports. + In the order of slave port index. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +oneOf: + - required: + - powerdown-gpios + - required: + - reset-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8855.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8855.yaml new file mode 100644 index 000000000..00a6a05e9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8855.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8855.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8855 stereo smart speaker amplifier + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com> + - Prasad Kumpatla <prasad.kumpatla@oss.qualcomm.com> + +description: + WSA885X is a Qualcomm Aqstic stereo smart speaker amplifier. It uses a PCM + audio interface, an I2C control interface, and a Class-H amplifier path for + high efficiency, low output noise, and low idle power consumption. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: qcom,wsa8855 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + powerdown-gpios: + description: GPIO controlling the SD_N powerdown pin. + maxItems: 1 + + reset-gpios: + description: GPIO controlling the SD_N powerdown pin. + maxItems: 1 + + interrupts: + maxItems: 1 + + vdd-1p8-supply: true + + vdd-io-supply: true + + qcom,battery-config: + description: + Battery topology connected to the speaker amplifier. 1S indicates one + battery cell in series, while 2S indicates two battery cells in series. + $ref: /schemas/types.yaml#/definitions/string + default: 1s + enum: + - 1s + - 2s + +required: + - compatible + - reg + - '#sound-dai-cells' + - interrupts + - vdd-1p8-supply + - vdd-io-supply + +oneOf: + - required: + - powerdown-gpios + - required: + - reset-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + speaker@c { + compatible = "qcom,wsa8855"; + reg = <0x0c>; + #sound-dai-cells = <0>; + powerdown-gpios = <&tlmm 11 GPIO_ACTIVE_LOW>; + interrupt-parent = <&tlmm>; + interrupts = <77 IRQ_TYPE_EDGE_FALLING>; + vdd-1p8-supply = <&vreg_l2g_1p8>; + vdd-io-supply = <&vreg_l1g_1p2>; + qcom,battery-config = "1s"; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/realtek,alc203.yaml b/Documentation/devicetree/bindings/sound/realtek,alc203.yaml new file mode 100644 index 000000000..6b90788b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,alc203.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,alc203.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek ALC203 AC97 Audio Codec + +maintainers: + - Keguang Zhang <keguang.zhang@gmail.com> + +description: + ALC203 is a full duplex AC97 2.3 compatible stereo audio codec. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,alc203 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + audio-codec { + compatible = "realtek,alc203"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,alc5623.yaml b/Documentation/devicetree/bindings/sound/realtek,alc5623.yaml new file mode 100644 index 000000000..683c58c3e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,alc5623.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,alc5623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5621/ALC5623 Audio Codec + +maintainers: + - Mahdi Khosravi <mmk1776@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - realtek,alc5621 + - realtek,alc5623 + + reg: + maxItems: 1 + + add-ctrl: + description: + Default register value for Reg-40h, Additional Control Register. + If absent or zero, the register is left untouched. + $ref: /schemas/types.yaml#/definitions/uint32 + + jack-det-ctrl: + description: + Default register value for Reg-5Ah, Jack Detect Control Register. + If absent or zero, the register is left untouched. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,alc5623"; + reg = <0x1a>; + add-ctrl = <0x3700>; + jack-det-ctrl = <0x4810>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml b/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml new file mode 100644 index 000000000..fb05988ff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,alc5632.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5632 audio CODEC + +description: | + Pins on the device (for linking into audio routes): + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + +maintainers: + - Leon Romanovsky <leon@leon.nu> + +properties: + compatible: + const: realtek,alc5632 + + reg: + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,alc5632"; + reg = <0x1a>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml new file mode 100644 index 000000000..3c44ebf93 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1015 Mono Class D Audio Amplifier + +maintainers: + - Jack Yu <jack.yu@realtek.com> + +properties: + compatible: + enum: + - realtek,rt1015 + + reg: + maxItems: 1 + + realtek,power-up-delay-ms: + description: Set a delay time for flush work to be completed, + this value is adjustable depending on platform. + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml new file mode 100644 index 000000000..7dac9e6f7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015p.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt1015p codec + +maintainers: + - Tzung-Bi Shih <tzungbi@kernel.org> + +description: | + Rt1015p is a rt1015 variant which does not support I2C and + only supports S24, 48kHz, 64FS. + +properties: + compatible: + enum: + - realtek,rt1015p + - realtek,rt1019p + + sdb-gpios: + description: + GPIO used for shutdown control. + 0 means shut down; 1 means power on. + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + rt1015p: rt1015p { + compatible = "realtek,rt1015p"; + sdb-gpios = <&pio 175 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml new file mode 100644 index 000000000..5287e9c91 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1016.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Reaktek RT1016 Stereo Class D Audio Amplifier + +maintainers: + - oder_chiou@realtek.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt1016 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,rt1016"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1019.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1019.yaml new file mode 100644 index 000000000..adf5e38f4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1019.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1019.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1019 Mono Class-D Audio Amplifier + +maintainers: + - jack.yu@realtek.com + +properties: + compatible: + const: realtek,rt1019 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt1019: codec@28 { + compatible = "realtek,rt1019"; + reg = <0x28>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5514.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5514.yaml new file mode 100644 index 000000000..7fbf7739c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5514.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5514.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT5514 audio CODEC + +maintainers: + - Animesh Agarwal <animeshagarwal28@gmail.com> + +description: | + This device supports both I2C and SPI. + + Pins on the device (for linking into audio routes) for I2C: + * DMIC1L + * DMIC1R + * DMIC2L + * DMIC2R + * AMICL + * AMICR + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5514 + + reg: + maxItems: 1 + + clocks: + items: + - description: Master clock to the CODEC + + clock-names: + items: + - const: mclk + + interrupts: + maxItems: 1 + description: The interrupt number to the cpu. + + realtek,dmic-init-delay-ms: + description: Set the DMIC initial delay (ms) to wait it ready for I2C. + + spi-max-frequency: true + + wakeup-source: + type: boolean + description: Flag to indicate this device can wake system (suspend/resume). + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5514"; + reg = <0x57>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml new file mode 100644 index 000000000..981ebc39b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5575.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5575 audio CODEC + +maintainers: + - Oder Chiou <oder_chiou@realtek.com> + +description: + The device supports both I2C and SPI. I2C is mandatory, while SPI is + optional depending on the hardware configuration. SPI is used for + firmware loading if present. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5575 + + reg: + maxItems: 1 + + spi-parent: + description: + Optional phandle reference to the SPI controller used for firmware + loading. The argument specifies the chip select. + $ref: /schemas/types.yaml#/definitions/phandle-array + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + # I2C-only node + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5575"; + reg = <0x57>; + }; + }; + + # I2C + optional SPI node + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5575"; + reg = <0x57>; + spi-parent = <&spi0 0>; /* chip-select 0 */ + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml new file mode 100644 index 000000000..29071044c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$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 + + clocks: + items: + - description: Master clock to the CODEC + + clock-names: + items: + - const: mclk + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5631.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5631.yaml new file mode 100644 index 000000000..747a731c4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5631.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5631.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5631/RT5631 audio CODEC + +maintainers: + - Animesh Agarwal <animeshagarwal28@gmail.com> + +description: | + This device supports I2C only. + + 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 + +properties: + compatible: + enum: + - realtek,alc5631 + - realtek,rt5631 + + reg: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,alc5631"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml new file mode 100644 index 000000000..922fe161f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml @@ -0,0 +1,156 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5640.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT5640/RT5639 audio CODEC + +maintainers: + - Neil Armstrong <neil.armstrong@linaro.org> + +description: | + This device supports I2C only. + + 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 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - realtek,rt5640 + - realtek,rt5639 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + interrupts: + maxItems: 1 + description: The CODEC's interrupt output. + + realtek,in1-differential: + description: + Indicate MIC1 input is differential, rather than single-ended. + type: boolean + + realtek,in2-differential: + description: + Indicate MIC2 input is differential, rather than single-ended. + type: boolean + + realtek,in3-differential: + description: + Indicate MIC3 input is differential, rather than single-ended. + type: boolean + + realtek,lout-differential: + description: + Indicate LOUT output is differential, rather than single-ended. + type: boolean + + realtek,dmic1-data-pin: + description: Specify which pin to be used as DMIC1 data pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 is not used + - 1 # using IN2P pin as dmic1 data pin + - 2 # using GPIO3 pin as dmic1 data pin + + realtek,dmic2-data-pin: + description: Specify which pin to be used as DMIC2 data pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic2 is not used + - 1 # using IN2N pin as dmic2 data pin + - 2 # using GPIO4 pin as dmic2 data pin + + realtek,jack-detect-source: + description: The Jack Detect source. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # Jack Detect function 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 + - 7 # Use HDA header for jack-detect + + realtek,jack-detect-not-inverted: + description: + 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. + type: boolean + + realtek,over-current-threshold-microamp: + description: micbias over-current detection threshold in µA + enum: + - 600 + - 1500 + - 2000 + + realtek,over-current-scale-factor: + description: micbias over-current detection scale-factor + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # Scale current by 0.5 + - 1 # Scale current by 0.75 + - 2 # Scale current by 1.0 + - 3 # Scale current by 1.5 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,rt5640"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5645.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5645.yaml new file mode 100644 index 000000000..0a698798c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5645.yaml @@ -0,0 +1,131 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5645.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT5650/RT5645 audio CODEC + +maintainers: + - Animesh Agarwal <animeshagarwal28@gmail.com> + +description: | + This device supports I2C only. + + 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 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - realtek,rt5645 + - realtek,rt5650 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + description: The CODEC's interrupt output. + + avdd-supply: + description: Power supply for AVDD, providing 1.8V. + + cpvdd-supply: + description: Power supply for CPVDD, providing 1.8V. + + hp-detect-gpios: + description: + 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. + maxItems: 1 + + cbj-sleeve-gpios: + description: + A GPIO spec to control the external combo jack circuit to tie the + sleeve/ring2 contacts to the ground or floating. It could avoid some + electric noise from the active speaker jacks. + maxItems: 1 + + realtek,in2-differential: + description: + Indicate MIC2 input are differential, rather than single-ended. + type: boolean + + realtek,dmic1-data-pin: + description: Specify which pin to be used as DMIC1 data pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 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: + description: Specify which pin to be used as DMIC2 data pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 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: + description: The JD mode of rt5645/rt5650. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # rt5645/rt5650 JD function is not used + - 1 # Mode-0 (VDD=3.3V), two port jack detection + - 2 # Mode-1 (VDD=3.3V), one port jack detection + - 3 # Mode-2 (VDD=1.8V), one port jack detection + +required: + - compatible + - reg + - interrupts + - avdd-supply + - cpvdd-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,rt5650"; + reg = <0x1a>; + hp-detect-gpios = <&gpio 19 0>; + cbj-sleeve-gpios = <&gpio 20 0>; + interrupt-parent = <&gpio>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&avdd_reg>; + cpvdd-supply = <&cpvdd_supply>; + realtek,jd-mode = <3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5651.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5651.yaml new file mode 100644 index 000000000..dc4f2eef7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5651.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5651.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek RT5651 audio CODEC + +maintainers: + - Bard Liao <bardliao@realtek.com> + +description: > + This device supports I2C only. + + 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 + +allOf: + - $ref: /schemas/sound/dai-common.yaml# + +properties: + compatible: + const: realtek,rt5651 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + '#sound-dai-cells': + const: 0 + + realtek,in2-differential: + type: boolean + description: Indicate MIC2 input are differential, rather than single-ended. + + realtek,dmic-en: + type: boolean + description: Indicates DMIC is used. + + realtek,jack-detect-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Select jack-detect input pin. + enum: [1, 2, 3] + + realtek,jack-detect-not-inverted: + type: boolean + description: + 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: + description: Micbias over-current detection threshold in µA. + enum: [600, 1500, 2000] + + realtek,over-current-scale-factor: + $ref: /schemas/types.yaml#/definitions/uint32 + description: > + Micbias over-current detection scale factor: + + 0: scale current by 0.5 + 1: scale current by 0.75 + 2: scale current by 1.0 + 3: scale current by 1.5 + enum: [0, 1, 2, 3] + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,rt5651"; + reg = <0x1a>; + realtek,dmic-en; + realtek,in2-differential; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5659.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5659.yaml new file mode 100644 index 000000000..1100ffd9a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5659.yaml @@ -0,0 +1,129 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5659.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT5659/RT5658 audio CODEC + +maintainers: + - Animesh Agarwal <animeshagarwal28@gmail.com> + +description: | + This device supports I2C only. + + 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 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - realtek,rt5659 + - realtek,rt5658 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + realtek,dmic1-data-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 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 + description: Specify which pin to be used as DMIC1 data pin. + default: 0 + + realtek,dmic2-data-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 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 + description: Specify which pin to be used as DMIC2 data pin. + default: 0 + + realtek,jd-src: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # No JD is used + - 1 # using JD3 as JD source + - 2 # JD source for Intel HDA header + description: Specify which JD source be used. + default: 0 + + realtek,ldo1-en-gpios: + maxItems: 1 + description: CODEC's LDO1_EN pin. + + realtek,reset-gpios: + maxItems: 1 + description: CODEC's RESET pin. + + ports: + $ref: /schemas/graph.yaml#/properties/ports + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1b { + compatible = "realtek,rt5659"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = <&gpio 3 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5677.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5677.yaml new file mode 100644 index 000000000..ae27ae78b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5677.yaml @@ -0,0 +1,135 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5677.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT5677 audio CODEC + +maintainers: + - Animesh Agarwal <animeshagarwal28@gmail.com> + +description: | + This device supports I2C only. + + Pins on the device (for linking into audio routes): + * IN1P + * IN1N + * IN2P + * IN2N + * MICBIAS1 + * DMIC1 + * DMIC2 + * DMIC3 + * DMIC4 + * LOUT1 + * LOUT2 + * LOUT3 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5677 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + gpio-controller: true + + '#gpio-cells': + const: 2 + + realtek,pow-ldo2-gpio: + maxItems: 1 + description: CODEC's POW_LDO2 pin. + + realtek,reset-gpio: + maxItems: 1 + description: CODEC's RESET pin. Active low. + + realtek,gpio-config: + description: | + Array of six 8bit elements that configures GPIO. + 0 - floating (reset value) + 1 - pull down + 2 - pull up + $ref: /schemas/types.yaml#/definitions/uint8-array + minItems: 6 + maxItems: 6 + items: + maximum: 2 + + realtek,jd1-gpio: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # OFF + - 1 # GPIO1 for jd1. + - 2 # GPIO2 for jd1. + - 3 # GPIO3 for jd1. + description: Configures GPIO Mic Jack detection 1. + + realtek,jd2-gpio: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # OFF + - 1 # GPIO4 for jd2. + - 2 # GPIO5 for jd2. + - 3 # GPIO6 for jd2. + description: Configures GPIO Mic Jack detection 2. + + realtek,jd3-gpio: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # OFF + - 1 # GPIO4 for jd3. + - 2 # GPIO5 for jd3. + - 3 # GPIO6 for jd3. + description: Configures GPIO Mic Jack detection 3. + +patternProperties: + '^realtek,in[1-2]-differential$': + type: boolean + description: Indicate MIC1/2 input are differential, rather than + single-ended. + + '^realtek,lout[1-3]-differential$': + type: boolean + description: Indicate LOUT1/2/3 outputs are differential, rather than + single-ended. + +required: + - compatible + - reg + - interrupts + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@2c { + compatible = "realtek,rt5677"; + reg = <0x2c>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_HIGH>; + gpio-controller; + #gpio-cells = <2>; + realtek,pow-ldo2-gpio = <&gpio 3 GPIO_ACTIVE_HIGH>; + realtek,reset-gpio = <&gpio 3 GPIO_ACTIVE_LOW>; + realtek,in1-differential; + realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682.yaml new file mode 100644 index 000000000..39333ea05 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682.yaml @@ -0,0 +1,156 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5682.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5682 and rt5682i codecs + +maintainers: + - Bard Liao <bardliao@realtek.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - realtek,rt5682 + - realtek,rt5682i + + 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 # using GPIO1 pin as dmic1 clock pin + - 1 # 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,btndet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + 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. + + 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,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. + + VBAT-supply: + description: Regulator supplying battery power through the VBAT 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. + +required: + - compatible + - reg + - AVDD-supply + - VBAT-supply + - MICVDD-supply + - DBVDD-supply + - LDO1-IN-supply + +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 { + compatible = "realtek,rt5682"; + 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>; + VBAT-supply = <&vbat_reg>; + MICVDD-supply = <&micvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + LDO1-IN-supply = <&ldo1_in_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml new file mode 100644 index 000000000..ecfa7a576 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -0,0 +1,150 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5682s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5682s codec + +maintainers: + - Derek Fang <derek.fang@realtek.com> + +description: | + Rt5682s(ALC5682I-VS) is a rt5682i variant which supports I2C only. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5682s + + reg: + maxItems: 1 + description: I2C address of the device. + + interrupts: + maxItems: 1 + description: The CODEC's interrupt output. + + realtek,dmic1-data-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 data is not used + - 1 # using GPIO2 pin as dmic1 data pin + - 2 # using GPIO5 pin as dmic1 data pin + description: | + Specify which GPIO pin be used as DMIC1 data pin. + + realtek,dmic1-clk-pin: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # dmic1 clk is not used + - 1 # using GPIO1 pin as dmic1 clock pin + - 2 # using GPIO3 pin as dmic1 clock pin + description: | + Specify which GPIO pin be used as DMIC1 clk pin. + + realtek,jd-src: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # No JD is used + - 1 # using JD1 as JD source + description: | + Specify which JD source be used. + + realtek,ldo1-en-gpios: + description: | + The GPIO that controls the CODEC's LDO1_EN pin. + + realtek,dmic-clk-rate-hz: + description: | + Set the clock rate (hz) for the requirement of the particular DMIC. + + realtek,dmic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular DMIC. + + realtek,amic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular platform or AMIC. + + realtek,dmic-clk-driving-high: + type: boolean + description: | + Set the high driving of the DMIC clock out. + + clocks: + items: + - description: phandle and clock specifier for codec MCLK. + + clock-names: + items: + - const: mclk + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: Name given for DAI word clock and bit clock outputs. + + "#sound-dai-cells": + const: 1 + + AVDD-supply: + description: Regulator supplying analog power through the AVDD pin. + + MICVDD-supply: + description: Regulator supplying power for the microphone bias through the + MICVDD pin. + + DBVDD-supply: + description: Regulator supplying I/O power through the DBVDD pin. + + LDO1-IN-supply: + description: Regulator supplying power to the digital core and charge pump + through the LDO1_IN pin. + +unevaluatedProperties: false + +required: + - compatible + - reg + - AVDD-supply + - MICVDD-supply + - DBVDD-supply + - LDO1-IN-supply + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,rt5682s"; + reg = <0x1a>; + interrupts = <6 IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio 2 GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; + + AVDD-supply = <&avdd_reg>; + MICVDD-supply = <&micvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + LDO1-IN-supply = <&ldo1_in_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml new file mode 100644 index 000000000..803945b7f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,fsi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas FIFO-buffered Serial Interface (FSI) + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for FSI2 SoC + - items: + - enum: + - renesas,fsi2-sh73a0 # SH-Mobile AG5 + - renesas,fsi2-r8a7740 # R-Mobile A1 + - enum: + - renesas,sh_fsi2 + # for Generic + - items: + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 1 + items: + - description: Main FSI module clock + - description: | + SPU bus/bridge clock. On R8A7740, this clock must be enabled to allow + register access as the FSI block is connected behind the SPU bus. + - description: CPG DIV6 functional clocks for FSI port A + - description: CPG DIV6 functional clocks for FSI port B + - description: FSI dividers for port A used for audio clock generation + - description: FSI dividers for port B used for audio clock generation + - description: External clock inputs for FSI port A provided by the board + - description: External clock inputs for FSI port B provided by the board + + clock-names: + minItems: 1 + maxItems: 8 + items: + enum: + - fck # Main FSI module clock + - spu # optional SPU bus/bridge clock + - icka # optional CPG DIV6 functional clocks for FSI port A + - ickb # optional CPG DIV6 functional clocks for FSI port B + - diva # optional FSI dividers for port A used for audio clock generation + - divb # optional FSI dividers for port B used for audio clock generation + - xcka # optional External clock inputs for FSI port A provided by the board + - xckb # optional External clock inputs for FSI port B provided by the board + + power-domains: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +patternProperties: + "^fsi(a|b),spdif-connection$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI is connected by S/PDIF + + "^fsi(a|b),stream-mode-support$": + $ref: /schemas/types.yaml#/definitions/flag + description: FSI supports 16bit stream mode + + "^fsi(a|b),use-internal-clock$": + $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 + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: renesas,fsi2-r8a7740 + then: + required: + - clock-names + + properties: + clock-names: + minItems: 2 + items: + - const: fck + - const: spu + - enum: [icka, ickb, diva, divb, xcka, xckb] + - enum: [icka, ickb, diva, divb, xcka, xckb] + - enum: [icka, ickb, diva, divb, xcka, xckb] + - enum: [icka, ickb, diva, divb, xcka, xckb] + - enum: [icka, ickb, diva, divb, xcka, xckb] + - enum: [icka, ickb, diva, divb, xcka, xckb] + +examples: + - | + #include <dt-bindings/clock/r8a7740-clock.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + sh_fsi2: sound@fe1f0000 { + compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2"; + reg = <0xfe1f0000 0x400>; + interrupts = <GIC_SPI 9 0x4>; + clocks = <&mstp3_clks R8A7740_CLK_FSI>, <&spu_clk>, + <&fsia_clk>, <&fsiack_clk>, <&fsidiva_clk>, + <&fsib_clk>, <&fsibck_clk>, <&fsidivb_clk>; + clock-names = "fck", "spu", "icka", "xcka", "diva", + "ickb", "xckb", "divb"; + power-domains = <&pd_a4mp>; + + #sound-dai-cells = <1>; + fsia,spdif-connection; + fsia,stream-mode-support; + fsia,use-internal-clock; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml b/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml new file mode 100644 index 000000000..a2b92dba5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,idt821034.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas IDT821034 codec device + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The IDT821034 codec is a four channel PCM codec with onchip filters and + programmable gain setting. + + The time-slots used by the codec must be set and so, the properties + 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and + 'dai-tdm-slot-rx-mask' must be present in the ALSA sound card node for + sub-nodes that involve the codec. The codec uses one 8bit time-slot per + channel. + 'dai-tdm-tdm-slot-with' must be set to 8. + + The IDT821034 codec also supports 5 gpios (SLIC signals) per channel. + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + - $ref: dai-common.yaml# + +properties: + compatible: + const: renesas,idt821034 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 8192000 + + spi-cpha: true + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - spi-cpha + - '#sound-dai-cells' + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + compatible = "renesas,idt821034"; + reg = <0>; + spi-max-frequency = <8192000>; + spi-cpha; + #sound-dai-cells = <0>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,r9a09g047-sound.yaml b/Documentation/devicetree/bindings/sound/renesas,r9a09g047-sound.yaml new file mode 100644 index 000000000..213c1c40b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,r9a09g047-sound.yaml @@ -0,0 +1,800 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,r9a09g047-sound.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas RZ/G3E Sound Controller + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + - John Madieu <john.madieu.xa@bp.renesas.com> + +description: + The RZ/G3E (R9A09G047) sound controller is based on R-Car Sound IP + with extended DMA channel support (up to 5 DMACs per direction), + additional clock domains (47 clocks including per-SSI ADG clocks), + and additional reset lines (14 including SCU, ADG and Audio DMAC + peri-peri resets). SSI operates exclusively in BUSIF mode with + 2-4 BUSIF channels per SSI. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: renesas,r9a09g047-sound + + reg: + maxItems: 5 + + reg-names: + items: + - const: scu + - const: adg + - const: ssiu + - const: ssi + - const: audmapp + + "#sound-dai-cells": + const: 1 + + "#clock-cells": + const: 0 + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + clocks: + maxItems: 47 + + clock-names: + items: + - const: ssi-all + - const: ssi-0 + - const: ssi-1 + - const: ssi-2 + - const: ssi-3 + - const: ssi-4 + - const: ssi-5 + - const: ssi-6 + - const: ssi-7 + - const: ssi-8 + - const: ssi-9 + - const: src-0 + - const: src-1 + - const: src-2 + - const: src-3 + - const: src-4 + - const: src-5 + - const: src-6 + - const: src-7 + - const: src-8 + - const: src-9 + - const: mix-0 + - const: mix-1 + - const: ctu-0 + - const: ctu-1 + - const: dvc-0 + - const: dvc-1 + - const: audio-clka + - const: audio-clkb + - const: audio-clkc + - const: audio-clki + - const: ssif_supply + - const: scu + - const: scu_x2 + - const: scu_supply + - const: adg-ssi-0 + - const: adg-ssi-1 + - const: adg-ssi-2 + - const: adg-ssi-3 + - const: adg-ssi-4 + - const: adg-ssi-5 + - const: adg-ssi-6 + - const: adg-ssi-7 + - const: adg-ssi-8 + - const: adg-ssi-9 + - const: audmapp + - const: adg + + power-domains: + maxItems: 1 + + resets: + maxItems: 14 + + reset-names: + items: + - const: ssi-all + - const: ssi-0 + - const: ssi-1 + - const: ssi-2 + - const: ssi-3 + - const: ssi-4 + - const: ssi-5 + - const: ssi-6 + - const: ssi-7 + - const: ssi-8 + - const: ssi-9 + - const: scu + - const: adg + - const: audmapp + + dvc: + type: object + additionalProperties: false + + patternProperties: + "^dvc-[0-1]$": + type: object + additionalProperties: false + + properties: + dmas: + maxItems: 5 + description: + List of references to DMA specifiers, one per DMA + controller, all for the transmission direction + (DVC is playback-only). The dma-engine core falls + through the list to find a free channel. + + dma-names: + maxItems: 5 + allOf: + - items: + enum: + - tx + + required: + - dmas + - dma-names + + mix: + type: object + additionalProperties: false + description: + Per-channel Mixer (MIX) sub-nodes. Each mix-N node has no + properties of its own. It exists so the driver can enumerate + the MIX instances and so that DT labels can be attached to it + for the dai/playback/capture phandle routing arrays. + + patternProperties: + "^mix-[0-1]$": + type: object + additionalProperties: false + + ctu: + type: object + additionalProperties: false + description: + Per-channel Channel Transfer Unit (CTU) sub-nodes. Each ctu-N + node has no properties of its own. It exists so the driver + can enumerate the CTU instances and so that DT labels can be + attached to it for the dai/playback/capture phandle routing arrays. + + patternProperties: + "^ctu-[0-7]$": + type: object + additionalProperties: false + + src: + type: object + additionalProperties: false + + patternProperties: + "^src-[0-9]$": + type: object + additionalProperties: false + + properties: + interrupts: + maxItems: 1 + + dmas: + maxItems: 10 + description: + Must contain a list of pairs of references to DMA + specifiers, one for transmission and one for reception, + repeated for each DMA controller. The dma-engine core + falls through the list to find a free channel. + + dma-names: + maxItems: 10 + allOf: + - items: + enum: + - tx + - rx + + ssiu: + type: object + additionalProperties: false + + patternProperties: + "^ssiu-[0-9]+$": + type: object + additionalProperties: false + + properties: + dmas: + maxItems: 10 + description: + Must contain a list of pairs of references to DMA + specifiers, one for transmission and one for reception, + repeated for each DMA controller. The dma-engine core + falls through the list to find a free channel. + + dma-names: + maxItems: 10 + allOf: + - items: + enum: + - tx + - rx + + required: + - dmas + - dma-names + + ssi: + type: object + additionalProperties: false + + patternProperties: + "^ssi-[0-9]$": + type: object + additionalProperties: false + + properties: + interrupts: + maxItems: 1 + + shared-pin: + description: Shared clock pin. + $ref: /schemas/types.yaml#/definitions/flag + + required: + - interrupts + + ports: + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + '^port@[0-9a-f]+$': + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + properties: + reg: + maxItems: 1 + endpoint: + $ref: audio-graph-port.yaml#/definitions/endpoint-base + unevaluatedProperties: false + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: + Ordered list of phandles to the in-SoC modules used + by this DAI in the playback direction. Each phandle + must reference one of the ssi-N, src-N, ctu-N, + mix-N or dvc-N sub-nodes of the parent sound + controller. The list order is the pipeline order + from CPU to off-SoC endpoint. + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: + Ordered list of phandles to the in-SoC modules used + by this DAI in the capture direction. Each phandle + must reference one of the ssi-N, src-N, ctu-N, + mix-N or dvc-N sub-nodes of the parent sound + controller. The list order is the pipeline order + from off-SoC endpoint to CPU. + +required: + - compatible + - reg + - reg-names + - "#sound-dai-cells" + - "#clock-cells" + - clocks + - clock-names + - resets + - reset-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + + sound@13c00000 { + #sound-dai-cells = <1>; + #clock-cells = <0>; + compatible = "renesas,r9a09g047-sound"; + reg = <0x13c00000 0x10000>, + <0x13c20000 0x10000>, + <0x13c30000 0x1000>, + <0x13c31000 0x1f000>, + <0x13c50000 0x10000>; + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + clocks = <&cpg 245>, + <&cpg 385>, <&cpg 386>, + <&cpg 387>, <&cpg 388>, + <&cpg 389>, <&cpg 390>, + <&cpg 391>, <&cpg 392>, + <&cpg 393>, <&cpg 394>, + <&cpg 372>, <&cpg 373>, + <&cpg 374>, <&cpg 375>, + <&cpg 376>, <&cpg 377>, + <&cpg 378>, <&cpg 379>, + <&cpg 380>, <&cpg 381>, + <&cpg 370>, <&cpg 371>, + <&cpg 370>, <&cpg 371>, + <&cpg 368>, <&cpg 369>, + <&cpg 251>, <&cpg 252>, + <&cpg 253>, <&cpg 250>, + <&cpg 384>, + <&cpg 246>, <&cpg 247>, + <&cpg 382>, + <&cpg 352>, <&cpg 353>, + <&cpg 354>, <&cpg 355>, + <&cpg 356>, <&cpg 357>, + <&cpg 358>, <&cpg 359>, + <&cpg 360>, <&cpg 361>, + <&cpg 248>, <&cpg 249>; + clock-names = "ssi-all", + "ssi-0", "ssi-1", + "ssi-2", "ssi-3", + "ssi-4", "ssi-5", + "ssi-6", "ssi-7", + "ssi-8", "ssi-9", + "src-0", "src-1", + "src-2", "src-3", + "src-4", "src-5", + "src-6", "src-7", + "src-8", "src-9", + "mix-0", "mix-1", + "ctu-0", "ctu-1", + "dvc-0", "dvc-1", + "audio-clka", "audio-clkb", + "audio-clkc", "audio-clki", + "ssif_supply", + "scu", "scu_x2", + "scu_supply", + "adg-ssi-0", "adg-ssi-1", + "adg-ssi-2", "adg-ssi-3", + "adg-ssi-4", "adg-ssi-5", + "adg-ssi-6", "adg-ssi-7", + "adg-ssi-8", "adg-ssi-9", + "audmapp", "adg"; + power-domains = <&cpg>; + resets = <&cpg 225>, + <&cpg 226>, <&cpg 227>, + <&cpg 228>, <&cpg 229>, + <&cpg 230>, <&cpg 231>, + <&cpg 232>, <&cpg 233>, + <&cpg 234>, <&cpg 235>, + <&cpg 236>, <&cpg 238>, <&cpg 237>; + reset-names = "ssi-all", + "ssi-0", "ssi-1", + "ssi-2", "ssi-3", + "ssi-4", "ssi-5", + "ssi-6", "ssi-7", + "ssi-8", "ssi-9", + "scu", "adg", + "audmapp"; + + ctu { + ctu-0 { }; + ctu-1 { }; + ctu-2 { }; + ctu-3 { }; + ctu-4 { }; + ctu-5 { }; + ctu-6 { }; + ctu-7 { }; + }; + + dvc { + dvc0: dvc-0 { + dmas = <&dmac0 0x1db3>, <&dmac1 0x1db3>, + <&dmac2 0x1db3>, <&dmac3 0x1db3>, + <&dmac4 0x1db3>; + dma-names = "tx", "tx", "tx", "tx", "tx"; + }; + dvc1: dvc-1 { + dmas = <&dmac0 0x1db4>, <&dmac1 0x1db4>, + <&dmac2 0x1db4>, <&dmac3 0x1db4>, + <&dmac4 0x1db4>; + dma-names = "tx", "tx", "tx", "tx", "tx"; + }; + }; + + mix { + mix-0 { }; + mix-1 { }; + }; + + src { + src0: src-0 { + interrupts = <GIC_SPI 902 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1d9f>, <&dmac0 0x1da9>, + <&dmac1 0x1d9f>, <&dmac1 0x1da9>, + <&dmac2 0x1d9f>, <&dmac2 0x1da9>, + <&dmac3 0x1d9f>, <&dmac3 0x1da9>, + <&dmac4 0x1d9f>, <&dmac4 0x1da9>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src1: src-1 { + interrupts = <GIC_SPI 903 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da0>, <&dmac0 0x1daa>, + <&dmac1 0x1da0>, <&dmac1 0x1daa>, + <&dmac2 0x1da0>, <&dmac2 0x1daa>, + <&dmac3 0x1da0>, <&dmac3 0x1daa>, + <&dmac4 0x1da0>, <&dmac4 0x1daa>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-2 { + interrupts = <GIC_SPI 904 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da1>, <&dmac0 0x1dab>, + <&dmac1 0x1da1>, <&dmac1 0x1dab>, + <&dmac2 0x1da1>, <&dmac2 0x1dab>, + <&dmac3 0x1da1>, <&dmac3 0x1dab>, + <&dmac4 0x1da1>, <&dmac4 0x1dab>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-3 { + interrupts = <GIC_SPI 905 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da2>, <&dmac0 0x1dac>, + <&dmac1 0x1da2>, <&dmac1 0x1dac>, + <&dmac2 0x1da2>, <&dmac2 0x1dac>, + <&dmac3 0x1da2>, <&dmac3 0x1dac>, + <&dmac4 0x1da2>, <&dmac4 0x1dac>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-4 { + interrupts = <GIC_SPI 906 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da3>, <&dmac0 0x1dad>, + <&dmac1 0x1da3>, <&dmac1 0x1dad>, + <&dmac2 0x1da3>, <&dmac2 0x1dad>, + <&dmac3 0x1da3>, <&dmac3 0x1dad>, + <&dmac4 0x1da3>, <&dmac4 0x1dad>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-5 { + interrupts = <GIC_SPI 907 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da4>, <&dmac0 0x1dae>, + <&dmac1 0x1da4>, <&dmac1 0x1dae>, + <&dmac2 0x1da4>, <&dmac2 0x1dae>, + <&dmac3 0x1da4>, <&dmac3 0x1dae>, + <&dmac4 0x1da4>, <&dmac4 0x1dae>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-6 { + interrupts = <GIC_SPI 908 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da5>, <&dmac0 0x1daf>, + <&dmac1 0x1da5>, <&dmac1 0x1daf>, + <&dmac2 0x1da5>, <&dmac2 0x1daf>, + <&dmac3 0x1da5>, <&dmac3 0x1daf>, + <&dmac4 0x1da5>, <&dmac4 0x1daf>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-7 { + interrupts = <GIC_SPI 909 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da6>, <&dmac0 0x1db0>, + <&dmac1 0x1da6>, <&dmac1 0x1db0>, + <&dmac2 0x1da6>, <&dmac2 0x1db0>, + <&dmac3 0x1da6>, <&dmac3 0x1db0>, + <&dmac4 0x1da6>, <&dmac4 0x1db0>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-8 { + interrupts = <GIC_SPI 910 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da7>, <&dmac0 0x1db1>, + <&dmac1 0x1da7>, <&dmac1 0x1db1>, + <&dmac2 0x1da7>, <&dmac2 0x1db1>, + <&dmac3 0x1da7>, <&dmac3 0x1db1>, + <&dmac4 0x1da7>, <&dmac4 0x1db1>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + src-9 { + interrupts = <GIC_SPI 911 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac0 0x1da8>, <&dmac0 0x1db2>, + <&dmac1 0x1da8>, <&dmac1 0x1db2>, + <&dmac2 0x1da8>, <&dmac2 0x1db2>, + <&dmac3 0x1da8>, <&dmac3 0x1db2>, + <&dmac4 0x1da8>, <&dmac4 0x1db2>; + dma-names = "rx", "tx", "rx", "tx", "rx", "tx", + "rx", "tx", "rx", "tx"; + }; + }; + + ssi { + ssi-0 { + interrupts = <GIC_SPI 889 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi-1 { + interrupts = <GIC_SPI 890 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi-2 { + interrupts = <GIC_SPI 891 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi3: ssi-3 { + interrupts = <GIC_SPI 892 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi4: ssi-4 { + interrupts = <GIC_SPI 893 IRQ_TYPE_LEVEL_HIGH>; + shared-pin; + }; + ssi-5 { + interrupts = <GIC_SPI 894 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi-6 { + interrupts = <GIC_SPI 895 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi-7 { + interrupts = <GIC_SPI 896 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi-8 { + interrupts = <GIC_SPI 897 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi-9 { + interrupts = <GIC_SPI 898 IRQ_TYPE_LEVEL_HIGH>; + }; + }; + + ssiu { + ssiu-0 { + dmas = <&dmac0 0x1d61>, <&dmac0 0x1d62>, + <&dmac1 0x1d61>, <&dmac1 0x1d62>, + <&dmac2 0x1d61>, <&dmac2 0x1d62>, + <&dmac3 0x1d61>, <&dmac3 0x1d62>, + <&dmac4 0x1d61>, <&dmac4 0x1d62>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-1 { + dmas = <&dmac0 0x1d63>, <&dmac0 0x1d64>, + <&dmac1 0x1d63>, <&dmac1 0x1d64>, + <&dmac2 0x1d63>, <&dmac2 0x1d64>, + <&dmac3 0x1d63>, <&dmac3 0x1d64>, + <&dmac4 0x1d63>, <&dmac4 0x1d64>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-2 { + dmas = <&dmac0 0x1d65>, <&dmac0 0x1d66>, + <&dmac1 0x1d65>, <&dmac1 0x1d66>, + <&dmac2 0x1d65>, <&dmac2 0x1d66>, + <&dmac3 0x1d65>, <&dmac3 0x1d66>, + <&dmac4 0x1d65>, <&dmac4 0x1d66>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-3 { + dmas = <&dmac0 0x1d67>, <&dmac0 0x1d68>, + <&dmac1 0x1d67>, <&dmac1 0x1d68>, + <&dmac2 0x1d67>, <&dmac2 0x1d68>, + <&dmac3 0x1d67>, <&dmac3 0x1d68>, + <&dmac4 0x1d67>, <&dmac4 0x1d68>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-4 { + dmas = <&dmac0 0x1d69>, <&dmac0 0x1d6a>, + <&dmac1 0x1d69>, <&dmac1 0x1d6a>, + <&dmac2 0x1d69>, <&dmac2 0x1d6a>, + <&dmac3 0x1d69>, <&dmac3 0x1d6a>, + <&dmac4 0x1d69>, <&dmac4 0x1d6a>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-5 { + dmas = <&dmac0 0x1d6b>, <&dmac0 0x1d6c>, + <&dmac1 0x1d6b>, <&dmac1 0x1d6c>, + <&dmac2 0x1d6b>, <&dmac2 0x1d6c>, + <&dmac3 0x1d6b>, <&dmac3 0x1d6c>, + <&dmac4 0x1d6b>, <&dmac4 0x1d6c>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-6 { + dmas = <&dmac0 0x1d6d>, <&dmac0 0x1d6e>, + <&dmac1 0x1d6d>, <&dmac1 0x1d6e>, + <&dmac2 0x1d6d>, <&dmac2 0x1d6e>, + <&dmac3 0x1d6d>, <&dmac3 0x1d6e>, + <&dmac4 0x1d6d>, <&dmac4 0x1d6e>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-7 { + dmas = <&dmac0 0x1d6f>, <&dmac0 0x1d70>, + <&dmac1 0x1d6f>, <&dmac1 0x1d70>, + <&dmac2 0x1d6f>, <&dmac2 0x1d70>, + <&dmac3 0x1d6f>, <&dmac3 0x1d70>, + <&dmac4 0x1d6f>, <&dmac4 0x1d70>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-8 { + dmas = <&dmac0 0x1d71>, <&dmac0 0x1d72>, + <&dmac1 0x1d71>, <&dmac1 0x1d72>, + <&dmac2 0x1d71>, <&dmac2 0x1d72>, + <&dmac3 0x1d71>, <&dmac3 0x1d72>, + <&dmac4 0x1d71>, <&dmac4 0x1d72>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-9 { + dmas = <&dmac0 0x1d73>, <&dmac0 0x1d74>, + <&dmac1 0x1d73>, <&dmac1 0x1d74>, + <&dmac2 0x1d73>, <&dmac2 0x1d74>, + <&dmac3 0x1d73>, <&dmac3 0x1d74>, + <&dmac4 0x1d73>, <&dmac4 0x1d74>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-10 { + dmas = <&dmac0 0x1d75>, <&dmac0 0x1d76>, + <&dmac1 0x1d75>, <&dmac1 0x1d76>, + <&dmac2 0x1d75>, <&dmac2 0x1d76>, + <&dmac3 0x1d75>, <&dmac3 0x1d76>, + <&dmac4 0x1d75>, <&dmac4 0x1d76>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-11 { + dmas = <&dmac0 0x1d77>, <&dmac0 0x1d78>, + <&dmac1 0x1d77>, <&dmac1 0x1d78>, + <&dmac2 0x1d77>, <&dmac2 0x1d78>, + <&dmac3 0x1d77>, <&dmac3 0x1d78>, + <&dmac4 0x1d77>, <&dmac4 0x1d78>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-12 { + dmas = <&dmac0 0x1d79>, <&dmac0 0x1d7a>, + <&dmac1 0x1d79>, <&dmac1 0x1d7a>, + <&dmac2 0x1d79>, <&dmac2 0x1d7a>, + <&dmac3 0x1d79>, <&dmac3 0x1d7a>, + <&dmac4 0x1d79>, <&dmac4 0x1d7a>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-13 { + dmas = <&dmac0 0x1d7b>, <&dmac0 0x1d7c>, + <&dmac1 0x1d7b>, <&dmac1 0x1d7c>, + <&dmac2 0x1d7b>, <&dmac2 0x1d7c>, + <&dmac3 0x1d7b>, <&dmac3 0x1d7c>, + <&dmac4 0x1d7b>, <&dmac4 0x1d7c>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-14 { + dmas = <&dmac0 0x1d7d>, <&dmac0 0x1d7e>, + <&dmac1 0x1d7d>, <&dmac1 0x1d7e>, + <&dmac2 0x1d7d>, <&dmac2 0x1d7e>, + <&dmac3 0x1d7d>, <&dmac3 0x1d7e>, + <&dmac4 0x1d7d>, <&dmac4 0x1d7e>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-15 { + dmas = <&dmac0 0x1d7f>, <&dmac0 0x1d80>, + <&dmac1 0x1d7f>, <&dmac1 0x1d80>, + <&dmac2 0x1d7f>, <&dmac2 0x1d80>, + <&dmac3 0x1d7f>, <&dmac3 0x1d80>, + <&dmac4 0x1d7f>, <&dmac4 0x1d80>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-16 { + dmas = <&dmac0 0x1d81>, <&dmac0 0x1d82>, + <&dmac1 0x1d81>, <&dmac1 0x1d82>, + <&dmac2 0x1d81>, <&dmac2 0x1d82>, + <&dmac3 0x1d81>, <&dmac3 0x1d82>, + <&dmac4 0x1d81>, <&dmac4 0x1d82>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-17 { + dmas = <&dmac0 0x1d83>, <&dmac0 0x1d84>, + <&dmac1 0x1d83>, <&dmac1 0x1d84>, + <&dmac2 0x1d83>, <&dmac2 0x1d84>, + <&dmac3 0x1d83>, <&dmac3 0x1d84>, + <&dmac4 0x1d83>, <&dmac4 0x1d84>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-18 { + dmas = <&dmac0 0x1d85>, <&dmac0 0x1d86>, + <&dmac1 0x1d85>, <&dmac1 0x1d86>, + <&dmac2 0x1d85>, <&dmac2 0x1d86>, + <&dmac3 0x1d85>, <&dmac3 0x1d86>, + <&dmac4 0x1d85>, <&dmac4 0x1d86>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-19 { + dmas = <&dmac0 0x1d87>, <&dmac0 0x1d88>, + <&dmac1 0x1d87>, <&dmac1 0x1d88>, + <&dmac2 0x1d87>, <&dmac2 0x1d88>, + <&dmac3 0x1d87>, <&dmac3 0x1d88>, + <&dmac4 0x1d87>, <&dmac4 0x1d88>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-20 { + dmas = <&dmac0 0x1d89>, <&dmac0 0x1d8a>, + <&dmac1 0x1d89>, <&dmac1 0x1d8a>, + <&dmac2 0x1d89>, <&dmac2 0x1d8a>, + <&dmac3 0x1d89>, <&dmac3 0x1d8a>, + <&dmac4 0x1d89>, <&dmac4 0x1d8a>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-21 { + dmas = <&dmac0 0x1d8b>, <&dmac0 0x1d8c>, + <&dmac1 0x1d8b>, <&dmac1 0x1d8c>, + <&dmac2 0x1d8b>, <&dmac2 0x1d8c>, + <&dmac3 0x1d8b>, <&dmac3 0x1d8c>, + <&dmac4 0x1d8b>, <&dmac4 0x1d8c>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-22 { + dmas = <&dmac0 0x1d8d>, <&dmac0 0x1d8e>, + <&dmac1 0x1d8d>, <&dmac1 0x1d8e>, + <&dmac2 0x1d8d>, <&dmac2 0x1d8e>, + <&dmac3 0x1d8d>, <&dmac3 0x1d8e>, + <&dmac4 0x1d8d>, <&dmac4 0x1d8e>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-23 { + dmas = <&dmac0 0x1d8f>, <&dmac0 0x1d90>, + <&dmac1 0x1d8f>, <&dmac1 0x1d90>, + <&dmac2 0x1d8f>, <&dmac2 0x1d90>, + <&dmac3 0x1d8f>, <&dmac3 0x1d90>, + <&dmac4 0x1d8f>, <&dmac4 0x1d90>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-24 { + dmas = <&dmac0 0x1d91>, <&dmac0 0x1d92>, + <&dmac1 0x1d91>, <&dmac1 0x1d92>, + <&dmac2 0x1d91>, <&dmac2 0x1d92>, + <&dmac3 0x1d91>, <&dmac3 0x1d92>, + <&dmac4 0x1d91>, <&dmac4 0x1d92>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-25 { + dmas = <&dmac0 0x1d93>, <&dmac0 0x1d94>, + <&dmac1 0x1d93>, <&dmac1 0x1d94>, + <&dmac2 0x1d93>, <&dmac2 0x1d94>, + <&dmac3 0x1d93>, <&dmac3 0x1d94>, + <&dmac4 0x1d93>, <&dmac4 0x1d94>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-26 { + dmas = <&dmac0 0x1d95>, <&dmac0 0x1d96>, + <&dmac1 0x1d95>, <&dmac1 0x1d96>, + <&dmac2 0x1d95>, <&dmac2 0x1d96>, + <&dmac3 0x1d95>, <&dmac3 0x1d96>, + <&dmac4 0x1d95>, <&dmac4 0x1d96>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + ssiu-27 { + dmas = <&dmac0 0x1d97>, <&dmac0 0x1d98>, + <&dmac1 0x1d97>, <&dmac1 0x1d98>, + <&dmac2 0x1d97>, <&dmac2 0x1d98>, + <&dmac3 0x1d97>, <&dmac3 0x1d98>, + <&dmac4 0x1d97>, <&dmac4 0x1d98>; + dma-names = "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx", "tx", "rx"; + }; + }; + + ports { + #address-cells = <1>; + #size-cells = <0>; + port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint>; + dai-format = "i2s"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + playback = <&ssi3>, <&src1>, <&dvc1>; + capture = <&ssi4>, <&src0>, <&dvc0>; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt new file mode 100644 index 000000000..3f07b072d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -0,0 +1,255 @@ +Renesas R-Car sound + +============================================= +* Modules +============================================= + +Renesas R-Car and RZ/G sound is constructed from below modules +(for Gen2 or later) + + SCU : Sampling Rate Converter Unit + - SRC : Sampling Rate Converter + - CMD + - CTU : Channel Transfer Unit + - MIX : Mixer + - DVC : Digital Volume and Mute Function + SSIU : Serial Sound Interface Unit + SSI : Serial Sound Interface + +See detail of each module's channels, connection, limitation on datasheet + +============================================= +* Multi channel +============================================= + +Multi channel is supported by Multi-SSI, or TDM-SSI. + + Multi-SSI : 6ch case, you can use stereo x 3 SSI + TDM-SSI : 6ch case, you can use TDM + +============================================= +* Enable/Disable each modules +============================================= + +See datasheet to check SRC/CTU/MIX/DVC connect-limitation. +DT controls enabling/disabling module. +${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example. +This is example of + +Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] +Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] + +see "Example: simple sound card" + +You can use below. +${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. + + &src0 &ctu00 &mix0 &dvc0 &ssi0 + &src1 &ctu01 &mix1 &dvc1 &ssi1 + &src2 &ctu02 &ssi2 + &src3 &ctu03 &ssi3 + &src4 &ssi4 + &src5 &ctu10 &ssi5 + &src6 &ctu11 &ssi6 + &src7 &ctu12 &ssi7 + &src8 &ctu13 &ssi8 + &src9 &ssi9 + +============================================= +* SRC (Sampling Rate Converter) +============================================= + + [xx]Hz [yy]Hz + ------> [SRC] ------> + +SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes + + Asynchronous mode: input data / output data are based on different clocks. + you can use this mode on Playback / Capture + Synchronous mode: input data / output data are based on same clocks. + This mode will be used if system doesn't have its input clock, + for example digital TV case. + you can use this mode on Playback + +------------------ +** Asynchronous mode +------------------ + +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +see "Example: simple sound card for Asynchronous mode" + +------------------ +** Synchronous mode +------------------ + + > amixer set "SRC Out Rate" on + > aplay xxxx.wav + > amixer set "SRC Out Rate" 48000 + > amixer set "SRC Out Rate" 44100 + +============================================= +* CTU (Channel Transfer Unit) +============================================= + + [xx]ch [yy]ch + ------> [CTU] --------> + +CTU can convert [xx]ch to [yy]ch, or exchange 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/renesas/rcar/ctu.c + - comment of header + +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +see "Example: simple sound card for channel convert" + +Ex) Exchange output channel + Input -> Output + 1ch -> 0ch + 0ch -> 1ch + + example of using matrix + output 0ch = (input 0ch x 0) + (input 1ch x 1) + output 1ch = (input 0ch x 1) + (input 1ch x 0) + + amixer set "CTU Reset" on + amixer set "CTU Pass" 9,10 + amixer set "CTU SV0" 0,4194304 + amixer set "CTU SV1" 4194304,0 + + example of changing connection + amixer set "CTU Reset" on + amixer set "CTU Pass" 2,1 + +============================================= +* MIX (Mixer) +============================================= + +MIX merges 2 sounds path. You can see 2 sound interface on system, +and these sounds will be merged by MIX. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 yyyy.wav + +You need to use "simple-scu-audio-card" or "audio-graph-scu-card" for it. +Ex) + [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] + | + [MEM] -> [SRC2] -> [CTU03] -+ + +see "Example: simple sound card for MIXer" + +============================================= +* DVC (Digital Volume and Mute Function) +============================================= + +DVC controls Playback/Capture volume. + +Playback Volume + amixer set "DVC Out" 100% + +Capture Volume + amixer set "DVC In" 100% + +Playback Mute + amixer set "DVC Out Mute" on + +Capture Mute + amixer set "DVC In Mute" on + +Volume Ramp + amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + amixer set "DVC Out Ramp" on + aplay xxx.wav & + amixer set "DVC Out" 80% // Volume Down + amixer set "DVC Out" 100% // Volume Up + +============================================= +* SSIU (Serial Sound Interface Unit) +============================================= + +SSIU can avoid some under/over run error, because it has some buffer. +But you can't use it if SSI was PIO mode. +In DMA mode, you can select not to use SSIU by using "no-busif" via SSI. + +SSIU handles BUSIF which will be used for TDM Split mode. +This driver is assuming that audio-graph card will be used. + +TDM Split mode merges 4 sounds. You can see 4 sound interface on system, +and these sounds will be merged SSIU/SSI. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 xxxx.wav & + aplay -D plughw:0,2 xxxx.wav & + aplay -D plughw:0,3 xxxx.wav + + 2ch 8ch + [MEM] -> [SSIU 30] -+-> [SSIU 3] --> [Codec] + 2ch | + [MEM] -> [SSIU 31] -+ + 2ch | + [MEM] -> [SSIU 32] -+ + 2ch | + [MEM] -> [SSIU 33] -+ + +see "Example: simple sound card for TDM Split" + +============================================= +* SSI (Serial Sound Interface) +============================================= + +** PIO mode + +You can use PIO mode which is for connection check by using. +Note: The system will drop non-SSI modules in PIO mode +even though if DT is selecting other modules. + + &ssi0 { + pio-transfer + }; + +** DMA mode without SSIU + +You can use DMA without SSIU. +Note: under/over run, or noise are likely to occur + + &ssi0 { + no-busif; + }; + +** PIN sharing + +Each SSI can share WS pin. It is based on platform. +This is example if SSI1 want to share WS pin with SSI0 + + &ssi1 { + shared-pin; + }; + +** Multi-SSI + +You can use Multi-SSI. +This is example of SSI0/SSI1/SSI2 (= for 6ch) + +see "Example: simple sound card for Multi channel" + +** TDM-SSI + +You can use TDM with SSI. +This is example of TDM 6ch. +Driver can automatically switches TDM <-> stereo mode in this case. + +see "Example: simple sound card for TDM" diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 000000000..e8a2acb92 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,541 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - const: renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7742 # RZ/G1H + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - const: renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a774e1 # RZ/G2H + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77961 # R-Car M3-W+ + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - const: renesas,rcar_sound-gen3 + # for Gen4 SoC + - items: + - enum: + - renesas,rcar_sound-r8a779g0 # R-Car V4H + - renesas,rcar_sound-r8a779h0 # R-Car V4M + - const: renesas,rcar_sound-gen4 + # for Generic + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + - renesas,rcar_sound-gen4 + + reg: + minItems: 1 + maxItems: 5 + + reg-names: + minItems: 1 + maxItems: 5 + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + This is used on simple-audio-card + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + + clock-frequency: + description: for audio_clkout0/1/2/3 + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + minItems: 1 + maxItems: 11 + + reset-names: + minItems: 1 + maxItems: 11 + + clocks: + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + minItems: 1 + maxItems: 31 + + clock-names: + description: List of necessary clock names. + # details are defined below + + # ports is below + port: + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: audio-graph-port.yaml#/definitions/endpoint-base + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + unevaluatedProperties: false + + rcar_sound,dvc: + description: DVC subnode. + type: object + patternProperties: + "^dvc-[0-1]$": + type: object + additionalProperties: false + + properties: + dmas: + maxItems: 1 + dma-names: + const: tx + required: + - dmas + - dma-names + additionalProperties: false + + rcar_sound,mix: + description: MIX subnode. + type: object + patternProperties: + "^mix-[0-1]$": + type: object + additionalProperties: false + additionalProperties: false + + rcar_sound,ctu: + description: CTU subnode. + type: object + patternProperties: + "^ctu-[0-7]$": + type: object + additionalProperties: false + additionalProperties: false + + rcar_sound,src: + description: SRC subnode. + type: object + patternProperties: + "^src-[0-9]$": + type: object + additionalProperties: false + + properties: + interrupts: + maxItems: 1 + dmas: + maxItems: 2 + dma-names: + allOf: + - items: + enum: + - tx + - rx + additionalProperties: false + + rcar_sound,ssiu: + description: SSIU subnode. + type: object + patternProperties: + "^ssiu-[0-9]+$": + type: object + additionalProperties: false + + properties: + dmas: + maxItems: 2 + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + rcar_sound,ssi: + description: SSI subnode. + type: object + patternProperties: + "^ssi-[0-9]$": + type: object + additionalProperties: false + + properties: + interrupts: + maxItems: 1 + dmas: + minItems: 2 + maxItems: 4 + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + additionalProperties: false + +patternProperties: + # For DAI base + 'rcar_sound,dai(@[0-9a-f]+)?$': + description: DAI subnode. + type: object + patternProperties: + "^dai([0-9]+)?$": + type: object + additionalProperties: false + + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + + 'ports(@[0-9a-f]+)?$': + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + '^port(@[0-9a-f]+)?$': + $ref: "#/properties/port" + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + # -------------------- + # reg/reg-names + # -------------------- + # for Gen1 + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen1 + then: + properties: + reg: + maxItems: 3 + reg-names: + items: + enum: + - sru + - ssi + - adg + # for Gen2/Gen3 + - if: + properties: + compatible: + contains: + enum: + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + then: + properties: + reg: + minItems: 5 + reg-names: + items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + # for Gen4 + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + properties: + reg: + maxItems: 4 + reg-names: + items: + enum: + - adg + - ssiu + - ssi + - sdmc + + # -------------------- + # clock-names + # -------------------- + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + properties: + clock-names: + maxItems: 3 + items: + enum: + - ssi.0 + - ssiu.0 + - clkin + else: + properties: + clock-names: + minItems: 1 + maxItems: 31 + items: + oneOf: + - const: ssi-all + - pattern: '^ssi\.[0-9]$' + - pattern: '^src\.[0-9]$' + - pattern: '^mix\.[0-1]$' + - pattern: '^ctu\.[0-1]$' + - pattern: '^dvc\.[0-1]$' + - pattern: '^clk_(a|b|c|i)$' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/r8a7790-cpg-mssr.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/power/r8a7790-sysc.h> + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0xec500000 0x1000>, /* SCU */ + <0xec5a0000 0x100>, /* ADG */ + <0xec540000 0x1000>, /* SSIU */ + <0xec541000 0x280>, /* SSI */ + <0xec740000 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&cpg CPG_MOD 1005>, /* SSI-ALL */ + <&cpg CPG_MOD 1006>, <&cpg CPG_MOD 1007>, /* SSI9, SSI8 */ + <&cpg CPG_MOD 1008>, <&cpg CPG_MOD 1009>, /* SSI7, SSI6 */ + <&cpg CPG_MOD 1010>, <&cpg CPG_MOD 1011>, /* SSI5, SSI4 */ + <&cpg CPG_MOD 1012>, <&cpg CPG_MOD 1013>, /* SSI3, SSI2 */ + <&cpg CPG_MOD 1014>, <&cpg CPG_MOD 1015>, /* SSI1, SSI0 */ + <&cpg CPG_MOD 1022>, <&cpg CPG_MOD 1023>, /* SRC9, SRC8 */ + <&cpg CPG_MOD 1024>, <&cpg CPG_MOD 1025>, /* SRC7, SRC6 */ + <&cpg CPG_MOD 1026>, <&cpg CPG_MOD 1027>, /* SRC5, SRC4 */ + <&cpg CPG_MOD 1028>, <&cpg CPG_MOD 1029>, /* SRC3, SRC2 */ + <&cpg CPG_MOD 1030>, <&cpg CPG_MOD 1031>, /* SRC1, SRC0 */ + <&cpg CPG_MOD 1020>, <&cpg CPG_MOD 1021>, /* MIX1, MIX0 */ + <&cpg CPG_MOD 1020>, <&cpg CPG_MOD 1021>, /* CTU1, CTU0 */ + <&cpg CPG_MOD 1019>, <&cpg CPG_MOD 1018>, /* DVC0, DVC1 */ + <&audio_clk_a>, <&audio_clk_b>, /* CLKA, CLKB */ + <&audio_clk_c>, <&audio_clk_i>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "mix.1", "mix.0", + "ctu.1", "ctu.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + power-domains = <&sysc R8A7790_PD_ALWAYS_ON>; + + resets = <&cpg 1005>, + <&cpg 1006>, <&cpg 1007>, <&cpg 1008>, <&cpg 1009>, + <&cpg 1010>, <&cpg 1011>, <&cpg 1012>, <&cpg 1013>, + <&cpg 1014>, <&cpg 1015>; + reset-names = "ssi-all", + "ssi.9", "ssi.8", "ssi.7", "ssi.6", + "ssi.5", "ssi.4", "ssi.3", "ssi.2", + "ssi.1", "ssi.0"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <GIC_SPI 353 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <GIC_SPI 371 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5>, <&src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* assume audio-graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0>, <&src0>, <&dvc0>; + capture = <&ssi1>, <&src1>, <&dvc1>; + }; + }; + }; + + /* assume audio-graph */ + codec { + port { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml new file mode 100644 index 000000000..1394f7828 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.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/renesas,rz-ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2) + +maintainers: + - Biju Das <biju.das.jz@bp.renesas.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + items: + - enum: + - renesas,r9a07g043-ssi # RZ/G2UL and RZ/Five + - renesas,r9a07g044-ssi # RZ/G2{L,LC} + - renesas,r9a07g054-ssi # RZ/V2L + - renesas,r9a08g045-ssi # RZ/G3S + - renesas,r9a08g046-ssi # RZ/G3L + - 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 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: rt + + '#sound-dai-cells': + const: 0 + + port: + $ref: audio-graph-port.yaml#/definitions/port-base + description: Connection to controller providing I2S signals + +required: + - compatible + - reg + - interrupts + - interrupt-names + - clocks + - clock-names + - resets + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/r9a07g044-cpg.h> + + ssi0: ssi@10049c00 { + compatible = "renesas,r9a07g044-ssi", + "renesas,rz-ssi"; + reg = <0x10049c00 0x400>; + interrupts = <GIC_SPI 326 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 327 IRQ_TYPE_EDGE_RISING>, + <GIC_SPI 328 IRQ_TYPE_EDGE_RISING>; + interrupt-names = "int_req", "dma_rx", "dma_tx"; + clocks = <&cpg CPG_MOD R9A07G044_SSI0_PCLK2>, + <&cpg CPG_MOD R9A07G044_SSI0_PCLK_SFR>, + <&audio_clk1>, + <&audio_clk2>; + clock-names = "ssi", "ssi_sfr", "audio_clk1", "audio_clk2"; + power-domains = <&cpg>; + resets = <&cpg R9A07G044_SSI0_RST_M2_REG>; + dmas = <&dmac 0x2655>, + <&dmac 0x2656>; + dma-names = "tx", "rx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml new file mode 100644 index 000000000..a1242e8e0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rt9120.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/richtek,rt9120.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Richtek RT9120 Class-D audio amplifier + +maintainers: + - ChiYuan Huang <cy_huang@richtek.com> + +description: | + The RT9120 is a high efficiency, I2S-input, stereo audio power amplifier + delivering 2*20W into 8 Ohm BTL speaker loads. It supports the wide input + voltage range from 4.5V to 26.4V to meet the need on most common + applications like as TV, monitors. home entertainment, electronic music + equipment. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - richtek,rt9120 + + reg: + description: I2C device address + maxItems: 1 + + pwdnn-gpios: + description: GPIO used for power down, low active + maxItems: 1 + + dvdd-supply: + description: | + Supply for the default on DVDD power, voltage domain must be 3P3V or 1P8V + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - dvdd-supply + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + rt9120@1a { + compatible = "richtek,rt9120"; + reg = <0x1a>; + pwdnn-gpios = <&gpio26 2 0>; + dvdd-supply = <&vdd_io_reg>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rt9123.yaml b/Documentation/devicetree/bindings/sound/richtek,rt9123.yaml new file mode 100644 index 000000000..819ca0620 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rt9123.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/richtek,rt9123.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Richtek RT9123/RTQ9124 Audio Amplifier + +maintainers: + - ChiYuan Huang <cy_huang@richtek.com> + +description: + RT9123 is a 3.2W mono Class-D audio amplifier that features high efficiency + and performance with ultra-low quiescent current. The digital audio interface + support various formats, including I2S, left-justified, right-justified, and + TDM formats. + + RTQ9124 is an ultra-low output noise, digital input, mono-channel Class-D + power amplifier that supports a 2.1MHz switching frequency. It integrates + both DC and AC load diagnostics, as well as real-time load monitoring to + assess speaker condition. The device operates from 4.5V to 18V and delivers + up to 30W output power. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - richtek,rt9123 + - richtek,rtq9124 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + enable-gpios: + maxItems: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@5e { + compatible = "richtek,rt9123"; + reg = <0x5e>; + enable-gpios = <&gpio 26 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rt9123p.yaml b/Documentation/devicetree/bindings/sound/richtek,rt9123p.yaml new file mode 100644 index 000000000..693511dfd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rt9123p.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/richtek,rt9123p.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Richtek RT9123P Audio Amplifier + +maintainers: + - ChiYuan Huang <cy_huang@richtek.com> + +description: + RT9123P is a RT9123 variant which does not support I2C control. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - richtek,rt9123p + + '#sound-dai-cells': + const: 0 + + enable-gpios: + maxItems: 1 + + enable-delay-ms: + description: + Delay time for 'ENABLE' pin changes intended to make I2S clocks ready to + prevent speaker pop noise. The unit is in millisecond. + +required: + - compatible + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + amplifier { + compatible = "richtek,rt9123p"; + enable-gpios = <&gpio 26 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/richtek,rtq9128.yaml b/Documentation/devicetree/bindings/sound/richtek,rtq9128.yaml new file mode 100644 index 000000000..a12566398 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/richtek,rtq9128.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + + The RTQ9154 is the family series of RTQ9128. The major change is to modify + the package size. Beside this, whole functions are almost all the same. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - enum: + - richtek,rtq9128 + - items: + - enum: + - richtek,rtq9154 + - const: richtek,rtq9128 + + reg: + maxItems: 1 + + enable-gpios: + maxItems: 1 + + richtek,tdm-input-data2-select: + type: boolean + description: + By default, if TDM mode is used, TDM data input will select 'DATA1' pin + as the data source. This option will configure TDM data input source from + 'DATA1' to 'DATA2' pin. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + speaker@1a { + compatible = "richtek,rtq9128"; + reg = <0x1a>; + enable-gpios = <&gpio 26 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml new file mode 100644 index 000000000..9435f3954 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -0,0 +1,191 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,i2s-tdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S/TDM Controller + +description: + The Rockchip I2S/TDM Controller is a Time Division Multiplexed + audio interface found in various Rockchip SoCs, allowing up + to 8 channels of audio over a serial interface. + +maintainers: + - Nicolas Frattaroli <frattaroli.nicolas@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - rockchip,px30-i2s-tdm + - rockchip,rk1808-i2s-tdm + - rockchip,rk3308-i2s-tdm + - rockchip,rk3568-i2s-tdm + - rockchip,rk3588-i2s-tdm + - rockchip,rv1126-i2s-tdm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + maxItems: 2 + items: + enum: + - rx + - tx + + clocks: + minItems: 3 + items: + - description: clock for TX + - description: clock for RX + - description: AHB clock driving the interface + - description: + Parent clock for mclk_tx (only required when using mclk-calibrate) + - description: + Parent clock for mclk_rx (only required when using mclk-calibrate) + - description: + Clock for sample rates that are an integer multiple of 8000 + (only required when using mclk-calibrate) + - description: + Clock for sample rates that are an integer multiple of 11025 + (only required when using mclk-calibrate) + + clock-names: + minItems: 3 + items: + - const: mclk_tx + - const: mclk_rx + - const: hclk + - const: mclk_tx_src + - const: mclk_rx_src + - const: mclk_root0 + - const: mclk_root1 + + resets: + minItems: 1 + maxItems: 2 + description: resets for the tx and rx directions + + reset-names: + minItems: 1 + maxItems: 2 + items: + enum: + - tx-m + - rx-m + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + power-domains: + maxItems: 1 + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + rockchip,trcm-sync-tx-only: + type: boolean + description: Use TX BCLK/LRCK for both TX and RX. + + rockchip,trcm-sync-rx-only: + type: boolean + description: Use RX BCLK/LRCK for both TX and RX. + + "#sound-dai-cells": + const: 0 + + rockchip,i2s-rx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of I2S RX sdis to I2S data bus lines. + By default, they are mapped one-to-one. + rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0. + maxItems: 4 + items: + enum: [0, 1, 2, 3] + + rockchip,i2s-tx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of I2S TX sdos to I2S data bus lines. + By default, they are mapped one-to-one. + rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0. + maxItems: 4 + items: + enum: [0, 1, 2, 3] + + rockchip,io-multiplex: + description: + Specify that the GPIO lines on the I2S bus are multiplexed such that + the direction (input/output) needs to be dynamically adjusted. + type: boolean + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + - resets + - reset-names + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3568-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + i2s@fe410000 { + compatible = "rockchip,rk3568-i2s-tdm"; + reg = <0x0 0xfe410000 0x0 0x1000>; + interrupts = <GIC_SPI 53 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru MCLK_I2S1_8CH_TX>, <&cru MCLK_I2S1_8CH_RX>, + <&cru HCLK_I2S1_8CH>; + clock-names = "mclk_tx", "mclk_rx", "hclk"; + dmas = <&dmac1 3>, <&dmac1 2>; + dma-names = "rx", "tx"; + resets = <&cru SRST_M_I2S1_8CH_TX>, <&cru SRST_M_I2S1_8CH_RX>; + reset-names = "tx-m", "rx-m"; + rockchip,trcm-sync-tx-only; + rockchip,grf = <&grf>; + #sound-dai-cells = <0>; + pinctrl-names = "default"; + pinctrl-0 = + <&i2s1m0_sclktx + &i2s1m0_sclkrx + &i2s1m0_lrcktx + &i2s1m0_lrckrx + &i2s1m0_sdi0 + &i2s1m0_sdi1 + &i2s1m0_sdi2 + &i2s1m0_sdi3 + &i2s1m0_sdo0 + &i2s1m0_sdo1 + &i2s1m0_sdo2 + &i2s1m0_sdo3>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml new file mode 100644 index 000000000..ff9e40049 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.yaml @@ -0,0 +1,123 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,pdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip PDM controller + +description: + The Pulse Density Modulation Interface Controller (PDMC) is + a PDM interface controller and decoder that support PDM format. + It integrates a clock generator driving the PDM microphone + and embeds filters which decimate the incoming bit stream to + obtain most common audio rates. + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - rockchip,pdm + - rockchip,px30-pdm + - rockchip,rk1808-pdm + - rockchip,rk3308-pdm + - rockchip,rk3568-pdm + - rockchip,rv1126-pdm + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for PDM controller + - description: clock for PDM BUS + + clock-names: + items: + - const: pdm_clk + - const: pdm_hclk + + dmas: + maxItems: 1 + + dma-names: + items: + - const: rx + + power-domains: + maxItems: 1 + + resets: + items: + - description: reset for PDM controller + + reset-names: + items: + - const: pdm-m + + rockchip,path-map: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of PDM SDIx to PDM PATHx. + By default, they are mapped one-to-one. + maxItems: 4 + uniqueItems: true + items: + enum: [ 0, 1, 2, 3 ] + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3328-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + + pdm@ff040000 { + compatible = "rockchip,pdm"; + reg = <0x0 0xff040000 0x0 0x1000>; + interrupts = <GIC_SPI 82 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_PDM>, <&cru HCLK_PDM>; + clock-names = "pdm_clk", "pdm_hclk"; + dmas = <&dmac 16>; + dma-names = "rx"; + #sound-dai-cells = <0>; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pdmm0_clk + &pdmm0_sdi0 + &pdmm0_sdi1 + &pdmm0_sdi2 + &pdmm0_sdi3>; + pinctrl-1 = <&pdmm0_clk_sleep + &pdmm0_sdi0_sleep + &pdmm0_sdi1_sleep + &pdmm0_sdi2_sleep + &pdmm0_sdi3_sleep>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3036-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3036-codec.yaml new file mode 100644 index 000000000..7570cc137 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3036-codec.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3036-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip RK3036 internal codec + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: rockchip,rk3036-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + + clock-names: + items: + - const: acodec_pclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3036-cru.h> + acodec: audio-codec@20030000 { + compatible = "rockchip,rk3036-codec"; + reg = <0x20030000 0x4000>; + rockchip,grf = <&grf>; + clock-names = "acodec_pclk"; + clocks = <&cru ACLK_VCODEC>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt new file mode 100644 index 000000000..73577ac1b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt @@ -0,0 +1,36 @@ +ROCKCHIP RK3288 with HDMI and analog audio + +Required properties: +- compatible: "rockchip,rk3288-hdmi-analog" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the analog audio codec. +- rockchip,routing: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. For this driver the first string should always be + "Analog". + +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/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml new file mode 100644 index 000000000..2cf229a07 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.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/rockchip,rk3308-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip RK3308 Internal Codec + +description: | + This is the audio codec embedded in the Rockchip RK3308 + SoC. It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported + sampling rate is 192 kHz. + + It is connected internally to one out of a selection of the internal I2S + controllers. + + The RK3308 audio codec has 8 independent capture channels, but some + features work on stereo pairs called groups: + * grp 0 -- MIC1 / MIC2 + * grp 1 -- MIC3 / MIC4 + * grp 2 -- MIC5 / MIC6 + * grp 3 -- MIC7 / MIC8 + +maintainers: + - Luca Ceresoli <luca.ceresoli@bootlin.com> + +properties: + compatible: + const: rockchip,rk3308-codec + + reg: + maxItems: 1 + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + Phandle to the General Register Files (GRF) + + clocks: + items: + - description: clock for TX + - description: clock for RX + - description: AHB clock driving the interface + + clock-names: + items: + - const: mclk_tx + - const: mclk_rx + - const: hclk + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + resets: + maxItems: 1 + + reset-names: + items: + - const: codec + + "#sound-dai-cells": + const: 0 + + rockchip,micbias-avdd-percent: + description: | + Voltage setting for the MICBIAS pins expressed as a percentage of + AVDD. + + E.g. if rockchip,micbias-avdd-percent = 85 and AVDD = 3v3, then the + MIC BIAS voltage will be 3.3 V * 85% = 2.805 V. + + enum: [ 50, 55, 60, 65, 70, 75, 80, 85 ] + +required: + - compatible + - reg + - rockchip,grf + - clocks + - resets + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3308-cru.h> + + audio_codec: audio-codec@ff560000 { + compatible = "rockchip,rk3308-codec"; + reg = <0xff560000 0x10000>; + rockchip,grf = <&grf>; + clock-names = "mclk_tx", "mclk_rx", "hclk"; + clocks = <&cru SCLK_I2S2_8CH_TX_OUT>, + <&cru SCLK_I2S2_8CH_RX_OUT>, + <&cru PCLK_ACODEC>; + reset-names = "codec"; + resets = <&cru SRST_ACODEC_P>; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000..52e3f1f90 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.yaml new file mode 100644 index 000000000..e9d13695c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3399-gru-sound.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip with MAX98357A/RT5514/DA7219 codecs on GRU boards + +maintainers: + - Heiko Stuebner <heiko@sntech.de> + +properties: + compatible: + const: rockchip,rk3399-gru-sound + + rockchip,cpu: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles to the Rockchip CPU DAI controllers connected to codecs + minItems: 1 + items: + - items: + - description: Phandle to the Rockchip I2S controllers + - items: + - description: | + Phandle to the Rockchip SPDIF controller. Required when a + DisplayPort audio codec is referenced in rockchip,codec + + rockchip,codec: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandles of the audio codecs connected to the Rockchip CPU DAI + controllers + minItems: 1 + maxItems: 6 + items: + maxItems: 1 + + dmic-wakeup-delay-ms: + description: | + specify delay time (ms) for DMIC ready. + If this option is specified, a delay is required for DMIC to get ready + so that rt5514 can avoid recording before DMIC sends valid data + +required: + - compatible + - rockchip,cpu + - rockchip,codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0 &spdif>; + rockchip,codec = <&max98357a &rt5514 &da7219 &cdn_dp>; + dmic-wakeup-delay-ms = <20>; + }; + diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3576-sai.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3576-sai.yaml new file mode 100644 index 000000000..149da9a91 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3576-sai.yaml @@ -0,0 +1,144 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3576-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip Serial Audio Interface Controller + +description: + The Rockchip Serial Audio Interface (SAI) controller is a flexible audio + controller that implements the I2S, I2S/TDM and the PDM standards. + +maintainers: + - Nicolas Frattaroli <nicolas.frattaroli@collabora.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: rockchip,rk3576-sai + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + items: + - enum: [tx, rx] + - const: rx + + clocks: + items: + - description: master audio clock + - description: AHB clock driving the interface + + clock-names: + items: + - const: mclk + - const: hclk + + resets: + minItems: 1 + items: + - description: reset for the mclk domain + - description: reset for the hclk domain + + reset-names: + minItems: 1 + items: + - const: m + - const: h + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + rockchip,sai-rx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of the controller's SDI ports to actual input lanes, + as well as the number of input lanes. + rockchip,sai-rx-route = <3> would mean sdi3 is receiving from data0, and + that there is only one receiving lane. + This property's absence is to be understood as only one receiving lane + being used if the controller has capture capabilities. + maxItems: 4 + items: + minimum: 0 + maximum: 3 + + rockchip,sai-tx-route: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Defines the mapping of the controller's SDO ports to actual output lanes, + as well as the number of output lanes. + rockchip,sai-tx-route = <3> would mean sdo3 is sending to data0, and + that there is only one transmitting lane. + This property's absence is to be understood as only one transmitting lane + being used if the controller has playback capabilities. + maxItems: 4 + items: + minimum: 0 + maximum: 3 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/rockchip,rk3576-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/pinctrl/rockchip.h> + #include <dt-bindings/power/rockchip,rk3576-power.h> + #include <dt-bindings/reset/rockchip,rk3576-cru.h> + + bus { + #address-cells = <2>; + #size-cells = <2>; + sai1: sai@2a610000 { + compatible = "rockchip,rk3576-sai"; + reg = <0x0 0x2a610000 0x0 0x1000>; + interrupts = <GIC_SPI 188 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru MCLK_SAI1_8CH>, <&cru HCLK_SAI1_8CH>; + clock-names = "mclk", "hclk"; + dmas = <&dmac0 2>, <&dmac0 3>; + dma-names = "tx", "rx"; + power-domains = <&power RK3576_PD_AUDIO>; + resets = <&cru SRST_M_SAI1_8CH>, <&cru SRST_H_SAI1_8CH>; + reset-names = "m", "h"; + pinctrl-names = "default"; + pinctrl-0 = <&sai1m0_lrck + &sai1m0_sclk + &sai1m0_sdi0 + &sai1m0_sdo0 + &sai1m0_sdo1 + &sai1m0_sdo2 + &sai1m0_sdo3>; + rockchip,sai-tx-route = <3 1 2 0>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rockchip-audio-max98090.yaml b/Documentation/devicetree/bindings/sound/rockchip,rockchip-audio-max98090.yaml new file mode 100644 index 000000000..5351d5f02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rockchip-audio-max98090.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/rockchip,rockchip-audio-max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip audio complex with MAX98090 codec + +maintainers: + - Fabio Estevam <festevam@gmail.com> + +properties: + compatible: + const: rockchip,rockchip-audio-max98090 + + rockchip,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + rockchip,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the Rockchip I2S controller. + + rockchip,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the MAX98090 audio codec. + + rockchip,headset-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the external chip for jack detection. + + rockchip,hdmi-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the HDMI device for HDMI codec. + +required: + - compatible + - rockchip,model + - rockchip,i2s-controller + +allOf: + - if: + required: + - rockchip,audio-codec + then: + required: + - rockchip,headset-codec + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&max98090>; + rockchip,headset-codec = <&headsetcodec>; + }; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000..fcb01abff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,140 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt b/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt new file mode 100644 index 000000000..411a62b3f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt @@ -0,0 +1,17 @@ +ROCKCHIP with RT5645/RT5650 CODECS + +Required properties: +- compatible: "rockchip,rockchip-audio-rt5645" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the RT5645/RT5650 audio codec + +Example: + +sound { + compatible = "rockchip,rockchip-audio-rt5645"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&rt5645>; +}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..b174d7498 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,113 @@ +# 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 + - items: + - enum: + - rockchip,rk3576-spdif + - rockchip,rk3588-spdif + - const: rockchip,rk3568-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF controller + - description: clock for SPDIF bus + + 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 + + port: + $ref: /schemas/graph.yaml#/properties/port + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - "#sound-dai-cells" + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + then: + required: + - rockchip,grf + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000..5abcf92bc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro@katsuster.net> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt new file mode 100644 index 000000000..02d53b9aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1011.txt @@ -0,0 +1,42 @@ +RT1011 Mono Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1011". + +- reg : The I2C address of the device. This I2C address decide by + two input pins (ASEL1 and ASEL2). + ------------------------------------- + | ASEL2 | ASEL1 | Address | + ------------------------------------- + | 0 | 0 | 0x38 | + ------------------------------------- + | 0 | 1 | 0x39 | + ------------------------------------- + | 1 | 0 | 0x3a | + ------------------------------------- + | 1 | 1 | 0x3b | + ------------------------------------- + +Optional properties: + +- realtek,temperature_calib + u32. The temperature was measured while doing the calibration. Units: Celsius degree + +- realtek,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + +Pins on the device (for linking into audio routes) for RT1011: + + * SPO + +Example: + +rt1011: codec@38 { + compatible = "realtek,rt1011"; + reg = <0x38>; + realtek,temperature_calib = <25>; + realtek,r0_calib = <0x224050>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt1308.txt b/Documentation/devicetree/bindings/sound/rt1308.txt new file mode 100644 index 000000000..2d46084af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1308.txt @@ -0,0 +1,17 @@ +RT1308 audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1308". + +- reg : The I2C address of the device. + + +Example: + +rt1308: rt1308@10 { + compatible = "realtek,rt1308"; + reg = <0x10>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt new file mode 100644 index 000000000..791a1bd76 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -0,0 +1,33 @@ +RT274 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt274". + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + + +Pins on the device (for linking into audio routes) for RT274: + + * DMIC1 Pin + * DMIC2 Pin + * MIC + * LINE1 + * LINE2 + * HPO Pin + * SPDIF + * LINE3 + +Example: + +rt274: codec@1c { + compatible = "realtek,rt274"; + reg = <0x1c>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt new file mode 100644 index 000000000..30be5f921 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5660.txt @@ -0,0 +1,47 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt new file mode 100644 index 000000000..24a6dab28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -0,0 +1,60 @@ +RT5663 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5663". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +- avdd-supply: Power supply for AVDD, providing 1.8V. + +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + +Optional properties: + +- "realtek,dc_offset_l_manual" +- "realtek,dc_offset_r_manual" +- "realtek,dc_offset_l_manual_mic" +- "realtek,dc_offset_r_manual_mic" + Based on the different PCB layout, add the manual offset value to + compensate the DC offset for each L and R channel, and they are different + between headphone and headset. +- "realtek,impedance_sensing_num" + The matrix row number of the impedance sensing table. + If the value is 0, it means the impedance sensing is not supported. +- "realtek,impedance_sensing_table" + The matrix rows of the impedance sensing table are consisted by impedance + minimum, impedance 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/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt new file mode 100644 index 000000000..f6ca96b4c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -0,0 +1,68 @@ +RT5665/RT5666 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5665", "realtek,rt5666". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential +- realtek,in3-differential +- realtek,in4-differential + Boolean. Indicate MIC1/2/3/4 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO4 pin as dmic1 data pin + 2: using IN2N pin as dmic2 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using GPIO5 pin as dmic2 data pin + 2: using IN2P pin as dmic2 data pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5659/RT5658: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * IN4P + * IN4N + * HPOL + * HPOR + * LOUTL + * LOUTR + * MONOOUT + * PDML + * PDMR + +Example: + +rt5659 { + compatible = "realtek,rt5665"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt new file mode 100644 index 000000000..a2b7e9a2f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -0,0 +1,50 @@ +RT5668B audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5668b" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5668B: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5668 { + compatible = "realtek,rt5668b"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml new file mode 100644 index 000000000..5ea0819a2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -0,0 +1,153 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Aries audio complex with WM8994 codec + +maintainers: + - Jonathan Bakker <xc-racer2@live.ca> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 + + cpu: + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 2 + maxItems: 2 + description: | + phandles to the I2S controller and bluetooth codec, + in that order + required: + - sound-dai + + codec: + additionalProperties: false + type: object + properties: + sound-dai: + maxItems: 1 + description: phandle to the WM8994 CODEC + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + deprecated: true + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the WM8994's pins (as + documented in its binding), and the jacks on the board - + For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic, + or FM In + For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic, + or HeadsetMic + Deprecated, use audio-routing. + + extcon: + description: Extcon phandle for dock detection + + main-micbias-supply: + description: Supply for the micbias on the main mic + + headset-micbias-supply: + description: Supply for the micbias on the headset mic + + earpath-sel-gpios: + maxItems: 1 + description: GPIO for switching between tv-out and mic paths + + headset-detect-gpios: + maxItems: 1 + description: GPIO for detection of headset insertion + + headset-key-gpios: + maxItems: 1 + description: GPIO for detection of headset key press + + io-channels: + maxItems: 1 + description: IO channel to read micbias voltage for headset detection + + io-channel-names: + const: headset-detect + +required: + - compatible + - cpu + - codec + - audio-routing + - extcon + - main-micbias-supply + - headset-micbias-supply + - earpath-sel-gpios + - headset-detect-gpios + - headset-key-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,fascinate4g-wm8994"; + + model = "Fascinate4G"; + + extcon = <&fsa9480>; + + main-micbias-supply = <&main_micbias_reg>; + headset-micbias-supply = <&headset_micbias_reg>; + + earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>; + + io-channels = <&adc 3>; + io-channel-names = "headset-detect"; + headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>; + headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>; + + audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "LINE", "LINEOUT2N", + "LINE", "LINEOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + + "IN1RP", "Headset Mic", + "IN1RN", "Headset Mic"; + + pinctrl-names = "default"; + pinctrl-0 = <&headset_det &earpath_sel>; + + cpu { + sound-dai = <&i2s0>, <&bt_codec>; + }; + + codec { + sound-dai = <&wm8994>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,arndale.yaml b/Documentation/devicetree/bindings/sound/samsung,arndale.yaml new file mode 100644 index 000000000..9bc4585bb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,arndale.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,arndale.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Insignal Arndale boards audio complex + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + enum: + - samsung,arndale-alc5631 + - samsung,arndale-rt5631 + - samsung,arndale-wm1811 + + samsung,audio-codec: + description: Phandle to the audio codec. + $ref: /schemas/types.yaml#/definitions/phandle + + samsung,audio-cpu: + description: Phandle to the Samsung I2S controller. + $ref: /schemas/types.yaml#/definitions/phandle + + samsung,model: + description: The user-visible name of this sound complex. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - samsung,audio-codec + - samsung,audio-cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,arndale-rt5631"; + samsung,audio-cpu = <&i2s0>; + samsung,audio-codec = <&rt5631>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 000000000..69ddfd4af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + headset-mic-bias-supply: + description: Supply for the micbias on the Headset microphone + + fm-sel-gpios: + maxItems: 1 + description: GPIO pin for FM selection + + lineout-sel-gpios: + maxItems: 1 + description: GPIO pin for line out selection + + 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 + + samsung,headset-4pole-threshold-microvolt: + minItems: 2 + maxItems: 2 + description: + Array containing minimum and maximum IO channel value for 4-pole + (with microphone/button) headsets. If the IO channel value is + outside of this range, a 3-pole headset is assumed. + + samsung,headset-button-threshold-microvolt: + minItems: 3 + maxItems: 3 + description: | + Array of minimum (inclusive) IO channel values for headset button + detection, in order: "Media", "Volume Up" and "Volume Down". + +required: + - compatible + - cpu + - codec + - audio-routing + - mic-bias-supply + - submic-bias-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000..c3dea852c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,92 @@ +# 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 + + 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: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + +required: + - compatible + - cpu + - codec + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml new file mode 100644 index 000000000..ac151d3c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,smdk5250.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SMDK5250 audio complex with WM8994 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + const: samsung,smdk-wm8994 + + samsung,audio-codec: + description: Phandle to the audio codec. + $ref: /schemas/types.yaml#/definitions/phandle + + samsung,i2s-controller: + description: Phandle to the Samsung I2S controller. + $ref: /schemas/types.yaml#/definitions/phandle + +required: + - compatible + - samsung,audio-codec + - samsung,i2s-controller + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,smdk-wm8994"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&wm8994>; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml new file mode 100644 index 000000000..3d49aa4c9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,snow.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Google Snow audio complex with MAX9809x codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +properties: + compatible: + enum: + - google,snow-audio-max98090 + - google,snow-audio-max98091 + - google,snow-audio-max98095 + + codec: + type: object + additionalProperties: false + properties: + sound-dai: + description: List of phandles to the CODEC and HDMI IP nodes. + items: + - description: Phandle to the MAX98090, MAX98091 or MAX98095 CODEC. + - description: Phandle to the HDMI IP block node. + required: + - sound-dai + + cpu: + type: object + additionalProperties: false + properties: + sound-dai: + description: Phandle to the Samsung I2S controller. + maxItems: 1 + required: + - sound-dai + + samsung,audio-codec: + description: Phandle to the audio codec. + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + + samsung,i2s-controller: + description: Phandle to the Samsung I2S controller. + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + + samsung,model: + description: The user-visible name of this sound complex. + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - codec + - cpu + +additionalProperties: false + +examples: + - | + sound { + compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&max98095 0>, <&hdmi>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml new file mode 100644 index 000000000..985b7d29c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - Sylwester Nawrocki <s.nawrocki@samsung.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: samsung,tm2-audio + + audio-amplifier: + description: Phandle to the MAX98504 amplifier. + $ref: /schemas/types.yaml#/definitions/phandle + + audio-codec: + description: Phandles to the codecs. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: Phandle to the WM5110 audio codec. + - items: + - description: Phandle to the HDMI transmitter node. + + samsung,audio-routing: + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are the + WM5110's and MAX98504's pins and the jacks on the board: HP, SPK, Main + Mic, Sub Mic, Third Mic, Headset Mic. + Deprecated, use audio-routing. + deprecated: true + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + i2s-controller: + description: Phandles to the I2S controllers. + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: Phandle to I2S0 + - description: Unused + - items: + - description: Phandle to I2S1 + - description: Unused + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + +required: + - compatible + - audio-amplifier + - audio-codec + - audio-routing + - i2s-controller + - mic-bias-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 GPIO_ACTIVE_HIGH>; + model = "wm5110"; + audio-routing = "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "RCV", "HPOUT3L", + "RCV", "HPOUT3R"; + }; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000..f45f73b50 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,162 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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. + oneOf: + - enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + - tesla,fsd-i2s + - items: + - enum: + - samsung,exynos5433-i2s + - const: samsung,exynos7-i2s + + '#address-cells': + const: 1 + + '#size-cells': + const: 0 + + reg: + maxItems: 1 + + 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: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/serial-midi.yaml b/Documentation/devicetree/bindings/sound/serial-midi.yaml new file mode 100644 index 000000000..3b2f6dd5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/serial-midi.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause + +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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) + +allOf: + - $ref: /schemas/serial/serial-peripheral-props.yaml# + +properties: + compatible: + const: serial-midi + + current-speed: + description: Baudrate to set the serial port to when this MIDI device is opened. + default: 38400 + +required: + - compatible + +additionalProperties: false + +examples: + - | + serial { + midi { + compatible = "serial-midi"; + }; + }; + - | + serial { + midi { + compatible = "serial-midi"; + current-speed = <115200>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml new file mode 100644 index 000000000..5db1f989d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-amplifier.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Amplifier + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - dioo,dio2125 + - simple-audio-amplifier + + enable-gpios: + maxItems: 1 + + VCC-supply: + description: > + power supply for the device + + sound-name-prefix: true + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/meson8-gpio.h> + + analog-amplifier { + compatible = "simple-audio-amplifier"; + VCC-supply = <®ulator>; + enable-gpios = <&gpio GPIOH_3 0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000..9b1bda485 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$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. + + state-labels: + description: State of input line. default is "Input 1", "Input 2" + $ref: /schemas/types.yaml#/definitions/string-array + maxItems: 2 + + idle-state: + description: If present specifies the state when the mux is powered down + $ref: /schemas/mux/mux-controller.yaml#/properties/idle-state + + sound-name-prefix: true + +required: + - compatible + - mux-gpios + +additionalProperties: false + +examples: + - | + mux { + compatible = "simple-audio-mux"; + mux-gpios = <&gpio 3 0>; + state-labels = "Label_A", "Label_B"; + idle-state = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000..23310f303 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,561 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$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 + + 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 + $ref: tdm-slot.yaml# + + 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 + + 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: + $ref: "#/definitions/pin-switches" + simple-audio-card,hp-det-gpio: + deprecated: true + maxItems: 1 + simple-audio-card,hp-det-gpios: + maxItems: 1 + simple-audio-card,mic-det-gpio: + deprecated: true + maxItems: 1 + simple-audio-card,mic-det-gpios: + maxItems: 1 + +patternProperties: + "^simple-audio-card,cpu(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + "^simple-audio-card,plat(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" + + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": + 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: + $ref: "#/definitions/pin-switches" + hp-det-gpio: + deprecated: true + maxItems: 1 + hp-det-gpios: + maxItems: 1 + mic-det-gpio: + deprecated: true + maxItems: 1 + mic-det-gpios: + maxItems: 1 + + patternProperties: + "^cpu(-[0-9]+)?$": + $ref: "#/definitions/dai" + "^codec(-[0-9]+)?$": + $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/Documentation/devicetree/bindings/sound/sirf-audio-port.txt b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt new file mode 100644 index 000000000..1f66de3c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt @@ -0,0 +1,20 @@ +* SiRF SoC audio port + +Required properties: +- compatible: "sirf,audio-port" +- reg: Base address and size entries: +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +Example: + +audioport: audioport@b0040000 { + compatible = "sirf,audio-port"; + reg = <0xb0040000 0x10000>; + dmas = <&dmac1 3>, <&dmac1 8>; + dma-names = "rx", "tx"; +}; diff --git a/Documentation/devicetree/bindings/sound/sirf-audio.txt b/Documentation/devicetree/bindings/sound/sirf-audio.txt new file mode 100644 index 000000000..c88882ca3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-audio.txt @@ -0,0 +1,41 @@ +* SiRF atlas6 and prima2 internal audio codec and port based audio setups + +Required properties: +- compatible: "sirf,sirf-audio-card" +- sirf,audio-platform: phandle for the platform node +- sirf,audio-codec: phandle for the SiRF internal codec node + +Optional properties: +- hp-pa-gpios: Need to be present if the board need control external + headphone amplifier. +- spk-pa-gpios: Need to be present if the board need control external + speaker amplifier. +- hp-switch-gpios: Need to be present if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Ext Spk + * Line In + * Mic + +SiRF internal audio codec pins: + * HPOUTL + * HPOUTR + * SPKOUT + * Ext Mic + * Mic Bias + +Example: + +sound { + compatible = "sirf,sirf-audio-card"; + sirf,audio-codec = <&audiocodec>; + sirf,audio-platform = <&audioport>; + hp-pa-gpios = <&gpio 44 0>; + spk-pa-gpios = <&gpio 46 0>; + hp-switch-gpios = <&gpio 45 0>; +}; + diff --git a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml new file mode 100644 index 000000000..a48d040b0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml @@ -0,0 +1,197 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/snps,designware-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: DesignWare I2S controller + +maintainers: + - Jose Abreu <joabreu@synopsys.com> + +properties: + compatible: + oneOf: + - items: + - const: canaan,k210-i2s + - const: snps,designware-i2s + - enum: + - snps,designware-i2s + - starfive,jh7110-i2stx0 + - starfive,jh7110-i2stx1 + - starfive,jh7110-i2srx + + reg: + maxItems: 1 + + interrupts: + description: | + The interrupt line number for the I2S controller. Add this + parameter if the I2S controller that you are using does not + support DMA. + maxItems: 1 + + clocks: + items: + - description: Sampling rate reference clock + - description: APB clock + - description: Audio master clock + - description: Inner audio master clock source + - description: External audio master clock source + - description: Bit clock + - description: Left/right channel clock + - description: External bit clock + - description: External left/right channel clock + minItems: 1 + + clock-names: + items: + - const: i2sclk + - const: apb + - const: mclk + - const: mclk_inner + - const: mclk_ext + - const: bclk + - const: lrck + - const: bclk_ext + - const: lrck_ext + minItems: 1 + + resets: + items: + - description: Optional controller resets + - description: controller reset of Sampling rate + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + minItems: 1 + + dma-names: + items: + - const: tx + - const: rx + minItems: 1 + + starfive,syscon: + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: phandle to System Register Controller sys_syscon node. + - description: I2S-rx enabled control offset of SYS_SYSCONSAIF__SYSCFG register. + - description: I2S-rx enabled control mask + description: + The phandle to System Register Controller syscon node and the I2S-rx(ADC) + enabled control offset and mask of SYS_SYSCONSAIF__SYSCFG register. + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: canaan,k210-i2s + then: + properties: + "#sound-dai-cells": + const: 1 + else: + properties: + "#sound-dai-cells": + const: 0 + - if: + properties: + compatible: + contains: + const: snps,designware-i2s + then: + properties: + clocks: + maxItems: 1 + clock-names: + maxItems: 1 + resets: + maxItems: 1 + else: + properties: + resets: + minItems: 2 + maxItems: 2 + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2stx0 + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + minItems: 5 + maxItems: 5 + required: + - resets + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2stx1 + then: + properties: + clocks: + minItems: 9 + maxItems: 9 + clock-names: + minItems: 9 + maxItems: 9 + required: + - resets + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2srx + then: + properties: + clocks: + minItems: 9 + maxItems: 9 + clock-names: + minItems: 9 + maxItems: 9 + required: + - resets + - starfive,syscon + else: + properties: + starfive,syscon: false + +required: + - compatible + - reg + - clocks + - clock-names + +oneOf: + - required: + - dmas + - dma-names + - required: + - interrupts + +unevaluatedProperties: false + +examples: + - | + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x7ff90000 0x1000>; + clocks = <&scpi_i2sclk 0>; + clock-names = "i2sclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/soc-ac97link.txt b/Documentation/devicetree/bindings/sound/soc-ac97link.txt new file mode 100644 index 000000000..80152a87f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc-ac97link.txt @@ -0,0 +1,28 @@ +AC97 link bindings + +These bindings can be included within any other device node. + +Required properties: + - pinctrl-names: Has to contain following states to setup the correct + pinmuxing for the used gpios: + "ac97-running": AC97-link is active + "ac97-reset": AC97-link reset state + "ac97-warm-reset": AC97-link warm reset state + - ac97-gpios: List of gpio phandles with args in the order ac97-sync, + ac97-sdata, ac97-reset + + +Example: + +ssi { + ... + + pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset"; + pinctrl-0 = <&ac97link_running>; + pinctrl-1 = <&ac97link_running>; + pinctrl-2 = <&ac97link_reset>; + pinctrl-3 = <&ac97link_warm_reset>; + ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>; + + ... +}; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000..8600520d7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,102 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel@alsa-project.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: /schemas/types.yaml#/definitions/phandle + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels that + depends on SoC. + ld11-aio,ld20-aio: + 0: hdmi + 1: pcmin2 + 2: line + 3: hpcmout1 + 4: pcmout3 + 5: hiecout1 + 6: epcmout2 + 7: epcmout3 + 8: hieccompout1 + pxs2-aio: + 0: hdmi + 1: line + 2: aux + 3: hiecout1 + 4: iecout1 + 5: hieccompout1 + 6: ieccompout1 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +unevaluatedProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000..985277648 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel@alsa-project.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + description: | + Port number of DT node is specified by the following DAI channels. + 0: line1 + 1: hp + 2: line2 + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +unevaluatedProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/sophgo,cv1800b-codecs.yaml b/Documentation/devicetree/bindings/sound/sophgo,cv1800b-codecs.yaml new file mode 100644 index 000000000..7293a98e9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sophgo,cv1800b-codecs.yaml @@ -0,0 +1,46 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sophgo,cv1800b-codecs.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Sophgo CV1800B Internal ADC/DAC Codec + +maintainers: + - Anton D. Stavinskii <stavinsky@gmail.com> + +description: + Internal ADC and DAC audio codecs integrated in the Sophgo CV1800B SoC. + Codecs expose a single DAI and are intended to be connected + to an I2S/TDM controller via an ASoC machine driver. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - sophgo,cv1800b-sound-adc + - sophgo,cv1800b-sound-dac + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + audio-codec@300a100 { + compatible = "sophgo,cv1800b-sound-adc"; + reg = <0x0300a100 0x100>; + #sound-dai-cells = <0>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sophgo,cv1800b-i2s.yaml b/Documentation/devicetree/bindings/sound/sophgo,cv1800b-i2s.yaml new file mode 100644 index 000000000..f08362b0c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sophgo,cv1800b-i2s.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sophgo,cv1800b-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Sophgo CV1800B I2S/TDM controller + +maintainers: + - Anton D. Stavinskii <stavinsky@gmail.com> + +description: I2S/TDM controller found in CV1800B / Sophgo SG2002/SG2000 SoCs. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sophgo,cv1800b-i2s + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 2 + + clock-names: + items: + - const: i2s + - const: mclk + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + items: + - enum: [rx, tx] + - const: tx + +required: + - compatible + - reg + - clocks + - clock-names + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/sophgo,cv1800.h> + + i2s@4110000 { + compatible = "sophgo,cv1800b-i2s"; + reg = <0x04110000 0x10000>; + clocks = <&clk CLK_APB_I2S1>, <&clk CLK_SDMA_AUD1>; + clock-names = "i2s", "mclk"; + dmas = <&dmamux 2 1>, <&dmamux 3 1>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml new file mode 100644 index 000000000..721950f65 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -0,0 +1,34 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-card-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Board Sound Card Common Properties + +maintainers: + - Mark Brown <broonie@kernel.org> + +properties: + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + ignore-suspend-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of audio sound widgets which are marked ignoring system suspend. + Paths between these endpoints are still active over suspend of the main + application processor that the current operating system is running. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +required: + - model + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/sound-dai.yaml b/Documentation/devicetree/bindings/sound/sound-dai.yaml new file mode 100644 index 000000000..ff9036e43 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-dai.yaml @@ -0,0 +1,20 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface consumer + +maintainers: + - Rob Herring <robh@kernel.org> + +select: true + +properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A phandle plus args to digital audio interface provider(s) + +additionalProperties: true +... diff --git a/Documentation/devicetree/bindings/sound/spacemit,k1-i2s.yaml b/Documentation/devicetree/bindings/sound/spacemit,k1-i2s.yaml new file mode 100644 index 000000000..72723f067 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/spacemit,k1-i2s.yaml @@ -0,0 +1,138 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/spacemit,k1-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: SpacemiT K1/K3 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: + - Troy Mitchell <troy.mitchell@linux.spacemit.com> + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: spacemit,k3-i2s + then: + properties: + clocks: + minItems: 6 + clock-names: + minItems: 6 + else: + properties: + clocks: + maxItems: 4 + clock-names: + maxItems: 4 + +properties: + compatible: + enum: + - spacemit,k1-i2s + - spacemit,k3-i2s + + reg: + maxItems: 1 + + clocks: + oneOf: + - items: + - description: clock for I2S sysclk + - description: clock for I2S bclk + - description: clock for I2S bus + - description: clock for I2S controller + - items: + - description: clock for I2S sysclk + - description: clock for I2S bclk + - description: clock for I2S bus + - description: clock for I2S controller + - description: clock for I2S common sysclk + - description: clock for I2S common bclk + - items: + - description: clock for I2S sysclk + - description: clock for I2S bclk + - description: clock for I2S bus + - description: clock for I2S controller + - description: clock for I2S sysclk divider + - description: clock for I2S common sysclk + - description: clock for I2S common bclk + + clock-names: + oneOf: + - items: + - const: sysclk + - const: bclk + - const: bus + - const: func + - items: + - const: sysclk + - const: bclk + - const: bus + - const: func + - const: c_sysclk + - const: c_bclk + - items: + - const: sysclk + - const: bclk + - const: bus + - const: func + - const: sysclk_div + - const: c_sysclk + - const: c_bclk + + dmas: + minItems: 1 + maxItems: 2 + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + resets: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - resets + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/spacemit,k1-syscon.h> + i2s@d4026000 { + compatible = "spacemit,k1-i2s"; + reg = <0xd4026000 0x30>; + clocks = <&syscon_mpmu CLK_I2S_SYSCLK>, + <&syscon_mpmu CLK_I2S_BCLK>, + <&syscon_apbc CLK_SSPA0_BUS>, + <&syscon_apbc CLK_SSPA0>; + clock-names = "sysclk", "bclk", "bus", "func"; + dmas = <&pdma0 21>, <&pdma0 22>; + dma-names = "tx", "rx"; + resets = <&syscon_apbc RESET_SSPA0>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml b/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml new file mode 100644 index 000000000..c15c01bbb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sprd,pcm-platform.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Spreadtrum DMA platform + +maintainers: + - Orson Zhai <orsonzhai@gmail.com> + - Baolin Wang <baolin.wang7@gmail.com> + - Chunyan Zhang <zhang.lyra@gmail.com> + +properties: + compatible: + const: sprd,pcm-platform + + dmas: + maxItems: 10 + + dma-names: + items: + - const: normal_p_l + - const: normal_p_r + - const: normal_c_l + - const: normal_c_r + - const: voice_c + - const: fast_p + - const: loop_c + - const: loop_p + - const: voip_c + - const: voip_p + +required: + - compatible + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + platform { + compatible = "sprd,pcm-platform"; + dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, + <&agcp_dma 3 3>, <&agcp_dma 4 4>, + <&agcp_dma 5 5>, <&agcp_dma 6 6>, + <&agcp_dma 7 7>, <&agcp_dma 8 8>, + <&agcp_dma 9 9>, <&agcp_dma 10 10>; + dma-names = "normal_p_l", "normal_p_r", + "normal_c_l", "normal_c_r", + "voice_c", "fast_p", + "loop_c", "loop_p", + "voip_c", "voip_p"; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml b/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml new file mode 100644 index 000000000..3b66bedef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sprd,sc9860-mcdt.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Spreadtrum Multi-Channel Data Transfer controller + +description: + The Multi-channel data transfer controller is used for sound stream + transmission between the audio subsystem and other AP/CP subsystem. It + supports 10 DAC channels and 10 ADC channels, and each channel can be + configured with DMA mode or interrupt mode. + +maintainers: + - Orson Zhai <orsonzhai@gmail.com> + - Baolin Wang <baolin.wang7@gmail.com> + - Chunyan Zhang <zhang.lyra@gmail.com> + +properties: + compatible: + const: sprd,sc9860-mcdt + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + mcdt@41490000 { + compatible = "sprd,sc9860-mcdt"; + reg = <0x41490000 0x170>; + interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ssm4567.txt b/Documentation/devicetree/bindings/sound/ssm4567.txt new file mode 100644 index 000000000..ec3d9e700 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm4567.txt @@ -0,0 +1,15 @@ +Analog Devices SSM4567 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm4567" + - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND), + 0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open). + +Example: + + ssm4567: ssm4567@34 { + compatible = "adi,ssm4567"; + reg = <0x34>; + }; diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt new file mode 100644 index 000000000..266b6fc9f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -0,0 +1,101 @@ +STA32X audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta32x" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - Vdda-supply: regulator spec, providing 3.3V + - Vdd3-supply: regulator spec, providing 3.3V + - Vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is chosen. + This properties have to be specified as '/bits/ 8' values. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,fault-detect-recovery: + If present, fault detect recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM speed mode (384 kHz) will be used. + + - st,invalid-input-detect-mute: + If present, automatic invalid input detect mute is enabled. + +Example: + +codec: sta32x@38 { + compatible = "st,sta32x"; + reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 000000000..e3d84864e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,131 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is chosen. + This properties have to be specified as '/bits/ 8' values. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM speed mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If present, automatic invalid input detect mute is enabled. + + - st,activate-mute-output: + If present, a mute output will be activated in ase the volume will + reach a value lower than -76 dBFS. + + - st,bridge-immediate-off: + If present, the bridge will be switched off immediately after the + power-down-gpio goes low. Otherwise, the bridge will wait for 13 + million clock cycles to pass before shutting down. + + - st,noise-shape-dc-cut: + If present, the noise-shaping technique on the DC cutoff filter are + enabled. + + - st,powerdown-master-volume: + If present, the power-down pin and I2C power-down functions will + act on the master volume. Otherwise, the functions will act on the + mute commands. + + - st,powerdown-delay-divider: + If present, the bridge power-down time will be divided by the provided + value. If not specified, a divider of 1 will be used. Allowed values + are 1, 2, 4, 8, 16, 32, 64 and 128. + This property has to be specified as '/bits/ 8' value. + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt new file mode 100644 index 000000000..a6ffcdec6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -0,0 +1,164 @@ +STMicroelectronics sti ASoC cards + +The sti ASoC Sound Card can be used, for all sti SoCs using internal sti-sas +codec or external codecs. + +sti sound drivers allows to expose sti SoC audio interface through the +generic ASoC simple card. For details about sound card declaration please refer to +Documentation/devicetree/bindings/sound/simple-card.yaml. + +1) sti-uniperiph-dai: audio dai device. +--------------------------------------- + +Required properties: + - compatible: "st,stih407-uni-player-hdmi", "st,stih407-uni-player-pcm-out", + "st,stih407-uni-player-dac", "st,stih407-uni-player-spdif", + "st,stih407-uni-reader-pcm_in", "st,stih407-uni-reader-hdmi", + + - st,syscfg: phandle to boot-device system configuration registers + + - clock-names: name of the clocks listed in clocks property in the same order + + - reg: CPU DAI IP Base address and size entries, listed in same + order than the CPU_DAI properties. + + - reg-names: names of the mapped memory regions listed in regs property in + the same order. + + - interrupts: CPU_DAI interrupt line, listed in the same order than the + CPU_DAI properties. + + - dma: CPU_DAI DMA controller phandle and DMA request line, listed in the same + order than the CPU_DAI properties. + + - dma-names: identifier string for each DMA request line in the dmas property. + "tx" for "st,sti-uni-player" compatibility + "rx" for "st,sti-uni-reader" compatibility + +Required properties ("st,sti-uni-player" compatibility only): + - clocks: CPU_DAI IP clock source, listed in the same order than the + CPU_DAI properties. + +Optional properties: + - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for + external codecs connection. + + - pinctrl-names: should contain only one value - "default". + + - st,tdm-mode: to declare to set TDM mode for unireader and uniplayer IPs. + Only compartible with IPs in charge of the external I2S/TDM bus. + Should be declared depending on associated codec. + +Example: + + sti_uni_player1: sti-uni-player@8d81000 { + compatible = "st,stih407-uni-player-hdmi"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + dma-names = "tx"; + st,tdm-mode = <1>; + }; + + sti_uni_player2: sti-uni-player@8d82000 { + compatible = "st,stih407-uni-player-pcm-out"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_2>; + reg = <0x8D82000 0x158>; + interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>; + dmas = <&fdma0 4 0 1>; + dma-names = "tx"; + }; + + sti_uni_player3: sti-uni-player@8d85000 { + compatible = "st,stih407-uni-player-spdif"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_SPDIFF>; + reg = <0x8D85000 0x158>; + interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>; + dmas = <&fdma0 7 0 1>; + dma-names = "tx"; + }; + + sti_uni_reader1: sti-uni-reader@8d84000 { + compatible = "st,stih407-uni-reader-hdmi"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + reg = <0x8D84000 0x158>; + interrupts = <GIC_SPI 88 IRQ_TYPE_NONE>; + dmas = <&fdma0 6 0 1>; + dma-names = "rx"; + }; + +2) sti-sas-codec: internal audio codec IPs driver +------------------------------------------------- + +Required properties: + - compatible: "st,sti<chip>-sas-codec" . + Should be chip "st,stih416-sas-codec" or "st,stih407-sas-codec" + + - st,syscfg: phandle to boot-device system configuration registers. + + - pinctrl-0: SPDIF PIO description. + + - pinctrl-names: should contain only one value - "default". + +Example: + sti_sas_codec: sti-sas-codec { + compatible = "st,stih407-sas-codec"; + #sound-dai-cells = <1>; + st,reg_audio = <&syscfg_core>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdif_out >; + }; + +Example of audio card declaration: + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "sti audio card"; + + simple-audio-card,dai-link@0 { + /* DAC */ + format = "i2s"; + dai-tdm-slot-width = <32>; + cpu { + sound-dai = <&sti_uni_player2>; + }; + + codec { + sound-dai = <&sti_sasg_codec 1>; + }; + }; + simple-audio-card,dai-link@1 { + /* SPDIF */ + format = "left_j"; + cpu { + sound-dai = <&sti_uni_player3>; + }; + + codec { + sound-dai = <&sti_sasg_codec 0>; + }; + }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000..b4f44f9c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,139 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + - st,stm32mp25-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. + minItems: 2 + + clock-names: + items: + - const: pclk + - const: i2sclk + - const: x8k + - const: x11k + minItems: 2 + + 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 + + access-controllers: + minItems: 1 + maxItems: 2 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + const: st,stm32h7-i2s + + then: + properties: + clocks: + minItems: 4 + + clock-names: + minItems: 4 + + - if: + properties: + compatible: + contains: + const: st,stm32mp25-i2s + + then: + properties: + clocks: + maxItems: 2 + + clock-names: + maxItems: 2 + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + + /* assume audio-graph */ + port { + codec_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000..551edf39e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,227 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan@foss.st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + - st,stm32mp25-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + minItems: 1 + maxItems: 3 + + access-controllers: + minItems: 1 + maxItems: 2 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + 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 + + - if: + properties: + compatible: + contains: + const: st,stm32h7-sai + then: + 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 + + - if: + properties: + compatible: + contains: + const: st,stm32mp25-sai + then: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + + clock-names: + items: + - const: pclk + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai2: sai@4400b000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400b000 0x400>; + reg = <0x4400b000 0x4>, <0x4400b3f0 0x10>; + clocks = <&rcc SAI2>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "x8k", "x11k"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>; + pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>; + + sai2a: audio-controller@4400b004 { + #sound-dai-cells = <0>; + compatible = "st,stm32-sai-sub-a"; + reg = <0x4 0x1c>; + dmas = <&dmamux1 89 0x400 0x01>; + dma-names = "tx"; + clocks = <&rcc SAI2_K>; + clock-names = "sai_ck"; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 000000000..56c5738ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-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 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + access-controllers: + minItems: 1 + maxItems: 2 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml b/Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml new file mode 100644 index 000000000..e2b4db6aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/starfive,jh7110-pwmdac.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: StarFive JH7110 PWM-DAC Controller + +description: + The PWM-DAC Controller uses PWM square wave generators plus RC filters to + form a DAC for audio play in StarFive JH7110 SoC. This audio play controller + supports 16 bit audio format, up to 48K sampling frequency, up to left and + right dual channels. + +maintainers: + - Hal Feng <hal.feng@starfivetech.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: starfive,jh7110-pwmdac + + reg: + maxItems: 1 + + clocks: + items: + - description: PWMDAC APB + - description: PWMDAC CORE + + clock-names: + items: + - const: apb + - const: core + + resets: + maxItems: 1 + description: PWMDAC APB + + dmas: + maxItems: 1 + description: TX DMA Channel + + dma-names: + const: tx + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - resets + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + pwmdac@100b0000 { + compatible = "starfive,jh7110-pwmdac"; + reg = <0x100b0000 0x1000>; + clocks = <&syscrg 157>, + <&syscrg 158>; + clock-names = "apb", "core"; + resets = <&syscrg 96>; + dmas = <&dma 22>; + dma-names = "tx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml b/Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml new file mode 100644 index 000000000..abb373fbf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/starfive,jh7110-tdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: StarFive JH7110 TDM Controller + +description: | + The TDM Controller is a Time Division Multiplexed audio interface + integrated in StarFive JH7110 SoC, allowing up to 8 channels of + audio over a serial interface. The TDM controller can operate both + in master and slave mode. + +maintainers: + - Walker Chen <walker.chen@starfivetech.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - starfive,jh7110-tdm + + reg: + maxItems: 1 + + clocks: + items: + - description: TDM AHB Clock + - description: TDM APB Clock + - description: TDM Internal Clock + - description: TDM Clock + - description: Inner MCLK + - description: TDM External Clock + + clock-names: + items: + - const: tdm_ahb + - const: tdm_apb + - const: tdm_internal + - const: tdm + - const: mclk_inner + - const: tdm_ext + + resets: + items: + - description: tdm ahb reset line + - description: tdm apb reset line + - description: tdm core reset line + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - resets + - dmas + - dma-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + tdm@10090000 { + compatible = "starfive,jh7110-tdm"; + reg = <0x10090000 0x1000>; + clocks = <&syscrg 184>, + <&syscrg 185>, + <&syscrg 186>, + <&syscrg 187>, + <&syscrg 17>, + <&tdm_ext>; + clock-names = "tdm_ahb", "tdm_apb", + "tdm_internal", "tdm", + "mclk_inner", "tdm_ext"; + resets = <&syscrg 105>, + <&syscrg 107>, + <&syscrg 106>; + dmas = <&dma 20>, <&dma 21>; + dma-names = "rx","tx"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/storm.txt b/Documentation/devicetree/bindings/sound/storm.txt new file mode 100644 index 000000000..062a4c185 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/storm.txt @@ -0,0 +1,23 @@ +* Sound complex for Storm boards + +Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC +connected to a MAX98357A DAC via I2S. + +Required properties: + +- compatible : "google,storm-audio" +- cpu : Phandle of the CPU DAI +- codec : Phandle of the codec DAI + +Optional properties: + +- qcom,model : The user-visible name of this sound card. + +Example: + +sound { + compatible = "google,storm-audio"; + qcom,model = "ipq806x-storm"; + cpu = <&lpass_cpu>; + codec = <&max98357a>; +}; diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt new file mode 100644 index 000000000..7d851ae2b --- /dev/null +++ b/Documentation/devicetree/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/Documentation/devicetree/bindings/sound/tda7419.txt b/Documentation/devicetree/bindings/sound/tda7419.txt new file mode 100644 index 000000000..6b85ec38d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tda7419.txt @@ -0,0 +1,38 @@ +TDA7419 audio processor + +This device supports I2C only. + +Required properties: + +- compatible : "st,tda7419" +- reg : the I2C address of the device. +- vdd-supply : a regulator spec for the common power supply (8-10V) + +Optional properties: + +- st,mute-gpios : a GPIO spec for the MUTE pin. + +Pins on the device (for linking into audio routes): + + * SE3L + * SE3R + * SE2L + * SE2R + * SE1L + * SE1R + * DIFFL + * DIFFR + * MIX + * OUTLF + * OUTRF + * OUTLR + * OUTRR + * OUTSW + +Example: + +ap: tda7419@44 { + compatible = "st,tda7419"; + reg = <0x44>; + vdd-supply = <&vdd_9v0_reg>; +}; diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.yaml b/Documentation/devicetree/bindings/sound/tdm-slot.yaml new file mode 100644 index 000000000..457a899e8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tdm-slot.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/tdm-slot.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Time Division Multiplexing (TDM) Slot Parameters + +maintainers: + - Liam Girdwood <lgirdwood@gmail.com> + +select: false + +properties: + dai-tdm-slot-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Number of slots in use + + dai-tdm-slot-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Width, in bits, of each slot + + dai-tdm-idle-mode: + $ref: /schemas/types.yaml#/definitions/string + enum: + - none + - off + - zero + - pulldown + - hiz + - pullup + - drivehigh + description: Drive mode for inactive/idle TDM slots. For hardware that + implements .set_tdm_idle(). Optional. "None" represents undefined + behaviour and is the same as not setting this property. + +patternProperties: + '^dai-tdm-slot-[rt]x-mask$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Slot mask for active TDM slots. Optional. Drivers may + specify .xlate_tdm_slot_mask() to generate a slot mask dynamically. If + neither this property nor a driver-specific function are specified, the + default snd_soc_xlate_tdm_slot_mask() function will be used to generate + a mask. The first element of the array is slot 0 (LSB). Any nonzero + value will be treated as 1. + + '^dai-tdm-slot-[rt]x-idle-mask$': + $ref: /schemas/types.yaml#/definitions/uint32 + description: Idle slot mask. Optional. A bit being set to 1 indicates + that the corresponding TDM slot is inactive/idle. + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/test-component.yaml b/Documentation/devicetree/bindings/sound/test-component.yaml new file mode 100644 index 000000000..9c40a2122 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/test-component.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/test-component.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Test Component + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +properties: + compatible: + enum: + - test-cpu + - test-cpu-verbose + - test-cpu-verbose-dai + - test-cpu-verbose-component + - test-codec + - test-codec-verbose + - test-codec-verbose-dai + - test-codec-verbose-component + +required: + - compatible + +additionalProperties: true + +examples: + - | + test_cpu { + compatible = "test-cpu"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,ads117x.txt b/Documentation/devicetree/bindings/sound/ti,ads117x.txt new file mode 100644 index 000000000..7db19b508 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,ads117x.txt @@ -0,0 +1,11 @@ +Texas Intstruments ADS117x ADC + +Required properties: + + - compatible : "ti,ads1174" or "ti,ads1178" + +Example: + +ads1178 { + compatible = "ti,ads1178"; +}; diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml new file mode 100644 index 000000000..20ea5883b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml @@ -0,0 +1,139 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi <peter.ujfalusi@ti.com> +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments J721e Common Processor Board Audio Support + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: | + The audio support on the board is using pcm3168a codec connected to McASP10 + serializers in parallel setup. + The pcm3168a SCKI clock is sourced from j721e AUDIO_REFCLK2 pin. + In order to support 48KHz and 44.1KHz family of sampling rates the parent + clock for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and + PLL15 (for 44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via + different HSDIVIDER. + + Clocking setup for j721e: + 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + + 44.1KHz family: + PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + + Clocking setup for j7200: + 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP0_AUXCLK ---> McASP0.auxclk + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + +properties: + compatible: + enum: + - ti,j721e-cpb-audio + - ti,j7200-cpb-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + ti,cpb-mcasp: + description: phandle to McASP used on CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,cpb-codec: + description: phandle to the pcm3168a codec used on the CPB + $ref: /schemas/types.yaml#/definitions/phandle + + clocks: + minItems: 4 + maxItems: 6 + + clock-names: + minItems: 4 + maxItems: 6 + +required: + - compatible + - model + - ti,cpb-mcasp + - ti,cpb-codec + - clocks + - clock-names + +additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: ti,j721e-cpb-audio + + then: + properties: + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: Parent for CPB_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + - description: Parent for CPB_SCKI clock (for 44.1KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-mcasp-auxclk-44100 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + - const: cpb-codec-scki-44100 + + - if: + properties: + compatible: + contains: + const: ti,j7200-cpb-audio + + then: + properties: + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + +examples: + - |+ + sound { + compatible = "ti,j721e-cpb-audio"; + model = "j721e-cpb"; + + ti,cpb-mcasp = <&mcasp10>; + ti,cpb-codec = <&pcm3168a_1>; + + clocks = <&k3_clks 184 1>, + <&k3_clks 184 2>, <&k3_clks 184 4>, + <&k3_clks 157 371>, + <&k3_clks 157 400>, <&k3_clks 157 401>; + clock-names = "cpb-mcasp-auxclk", + "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100", + "cpb-codec-scki", + "cpb-codec-scki-48000", "cpb-codec-scki-44100"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml new file mode 100644 index 000000000..5b2874a80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi <peter.ujfalusi@ti.com> +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments J721e Common Processor Board Audio Support + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: | + The Infotainment board plugs into the Common Processor Board, the support of the + extension board is extending the CPB audio support, described in: + sound/ti,j721e-cpb-audio.txt + + The audio support on the Infotainment Expansion Board consists of McASP0 + connected to two pcm3168a codecs with dedicated set of serializers to each. + The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin. + + In order to support 48KHz and 44.1KHz family of sampling rates the parent clock + for AUDIO_REFCLK0 needs to be changed between PLL4 (for 48KHz) and PLL15 (for + 44.1KHz). The same PLLs are used for McASP0's AUXCLK clock via different + HSDIVIDER. + + Note: the same PLL4 and PLL15 is used by the audio support on the CPB! + + Clocking setup for 48KHz family: + PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + | |-> MCASP0_AUXCLK ---> McASP0.auxclk + | + |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI + + Clocking setup for 44.1KHz family: + PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk + | |-> MCASP0_AUXCLK ---> McASP0.auxclk + | + |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI + |-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI + +properties: + compatible: + items: + - const: ti,j721e-cpb-ivi-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + ti,cpb-mcasp: + description: phandle to McASP used on CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,cpb-codec: + description: phandle to the pcm3168a codec used on the CPB + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-mcasp: + description: phandle to McASP used on IVI + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-codec-a: + description: phandle to the pcm3168a-A codec on the expansion board + $ref: /schemas/types.yaml#/definitions/phandle + + ti,ivi-codec-b: + description: phandle to the pcm3168a-B codec on the expansion board + $ref: /schemas/types.yaml#/definitions/phandle + + clocks: + items: + - description: AUXCLK clock for McASP used by CPB audio + - description: Parent for CPB_McASP auxclk (for 48KHz) + - description: Parent for CPB_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on CPB + - description: Parent for CPB_SCKI clock (for 48KHz) + - description: Parent for CPB_SCKI clock (for 44.1KHz) + - description: AUXCLK clock for McASP used by IVI audio + - description: Parent for IVI_McASP auxclk (for 48KHz) + - description: Parent for IVI_McASP auxclk (for 44.1KHz) + - description: SCKI clock for the pcm3168a codec on IVI + - description: Parent for IVI_SCKI clock (for 48KHz) + - description: Parent for IVI_SCKI clock (for 44.1KHz) + + clock-names: + items: + - const: cpb-mcasp-auxclk + - const: cpb-mcasp-auxclk-48000 + - const: cpb-mcasp-auxclk-44100 + - const: cpb-codec-scki + - const: cpb-codec-scki-48000 + - const: cpb-codec-scki-44100 + - const: ivi-mcasp-auxclk + - const: ivi-mcasp-auxclk-48000 + - const: ivi-mcasp-auxclk-44100 + - const: ivi-codec-scki + - const: ivi-codec-scki-48000 + - const: ivi-codec-scki-44100 + +required: + - compatible + - model + - ti,cpb-mcasp + - ti,cpb-codec + - ti,ivi-mcasp + - ti,ivi-codec-a + - ti,ivi-codec-b + - clocks + - clock-names + +additionalProperties: false + +examples: + - |+ + sound { + compatible = "ti,j721e-cpb-ivi-audio"; + model = "j721e-cpb-ivi"; + + ti,cpb-mcasp = <&mcasp10>; + ti,cpb-codec = <&pcm3168a_1>; + + ti,ivi-mcasp = <&mcasp0>; + ti,ivi-codec-a = <&pcm3168a_a>; + ti,ivi-codec-b = <&pcm3168a_b>; + + clocks = <&k3_clks 184 1>, + <&k3_clks 184 2>, <&k3_clks 184 4>, + <&k3_clks 157 371>, + <&k3_clks 157 400>, <&k3_clks 157 401>, + <&k3_clks 174 1>, + <&k3_clks 174 2>, <&k3_clks 174 4>, + <&k3_clks 157 301>, + <&k3_clks 157 330>, <&k3_clks 157 331>; + clock-names = "cpb-mcasp-auxclk", + "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100", + "cpb-codec-scki", + "cpb-codec-scki-48000", "cpb-codec-scki-44100", + "ivi-mcasp-auxclk", + "ivi-mcasp-auxclk-48000", "ivi-mcasp-auxclk-44100", + "ivi-codec-scki", + "ivi-codec-scki-48000", "ivi-codec-scki-44100"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,omap-twl4030.yaml b/Documentation/devicetree/bindings/sound/ti,omap-twl4030.yaml new file mode 100644 index 000000000..27c7019bd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,omap-twl4030.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/ti,omap-twl4030.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SoC with twl4030 based audio setups + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: + Audio setups on TI OMAP SoCs using TWL4030-family + audio codec connected via a McBSP port. + +properties: + compatible: + const: ti,omap-twl4030 + + ti,model: + $ref: /schemas/types.yaml#/definitions/string + description: Name of the sound card (for example "omap3beagle"). + + ti,mcbsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle for the McBSP node. + + ti,codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle for the twl4030 audio node. + + ti,mcbsp-voice: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle to the McBSP node connected to the voice port. + + ti,jack-det-gpio: + description: GPIO specifier for jack detection. + maxItems: 1 + + ti,audio-routing: + description: | + A list of audio routing connections. Each entry is a pair of strings, + with the first being the connection's sink and the second being the + source. If not provided, all possible connections are available. + + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headset Stereophone + - Earpiece Spk + - Handsfree Spk + - Ext Spk + - Main Mic + - Sub Mic + - Headset Mic + - Carkit Mic + - Digital0 Mic + - Digital1 Mic + - Line In + + # CODEC Pins + - HSOL + - HSOR + - EARPIECE + - HFL + - HFR + - PREDRIVEL + - PREDRIVER + - CARKITL + - CARKITR + - MAINMIC + - SUBMIC + - HSMIC + - DIGIMIC0 + - DIGIMIC1 + - CARKITMIC + - AUXL + - AUXR + + # Headset Mic Bias + - Mic Bias 1 # Used for Main Mic or Digimic0 + - Mic Bias 2 # Used for Sub Mic or Digimic1 + +required: + - compatible + - ti,model + - ti,mcbsp + +additionalProperties: false + +examples: + - | + sound { + compatible = "ti,omap-twl4030"; + ti,model = "omap3beagle"; + ti,mcbsp = <&mcbsp2>; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,omap4-dmic.yaml b/Documentation/devicetree/bindings/sound/ti,omap4-dmic.yaml new file mode 100644 index 000000000..773b9bf2f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,omap4-dmic.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,omap4-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments OMAP4+ Digital Microphone Module + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@ti.com> + +properties: + compatible: + const: ti,omap4-dmic + + reg: + items: + - description: MPU access base address + - description: L3 interconnect address + + reg-names: + items: + - const: mpu + - const: dma + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + items: + - const: up_link + +required: + - compatible + - reg + - reg-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + + dmic@0 { + compatible = "ti,omap4-dmic"; + reg = <0x0 0x7f>, /* MPU private access */ + <0x4902e000 0x7f>; /* L3 Interconnect */ + reg-names = "mpu", "dma"; + interrupts = <GIC_SPI 114 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&sdma 67>; + dma-names = "up_link"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml b/Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml new file mode 100644 index 000000000..cdea0a008 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,omap4-mcpdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: OMAP McPDM + +maintainers: + - Misael Lopez Cruz <misael.lopez@ti.com> + +description: + OMAP ALSA SoC DAI driver using McPDM port used by TWL6040 + +properties: + compatible: + const: ti,omap4-mcpdm + + reg: + items: + - description: MPU access base address + - description: L3 interconnect address + + reg-names: + items: + - const: mpu + - const: dma + + interrupts: + maxItems: 1 + + dmas: + maxItems: 2 + + dma-names: + items: + - const: up_link + - const: dn_link + + clocks: + maxItems: 1 + + clock-names: + items: + - const: pdmclk + +required: + - compatible + - reg + - reg-names + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + mcpdm@0 { + compatible = "ti,omap4-mcpdm"; + reg = <0x0 0x7f>, /* MPU private access */ + <0x49032000 0x7f>; /* L3 Interconnect */ + reg-names = "mpu", "dma"; + interrupts = <GIC_SPI 112 IRQ_TYPE_LEVEL_HIGH>; + interrupt-parent = <&gic>; + dmas = <&sdma 65>, <&sdma 66>; + dma-names = "up_link", "dn_link"; + clocks = <&twl6040>; + clock-names = "pdmclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml new file mode 100644 index 000000000..1f0e6787a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1681 8-channel Digital-to-Analog Converter + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + - Kevin Lu <kevin-lu@ti.com> + - Baojun Xu <baojun.xu@ti.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: ti,pcm1681 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm1681: audio-codec@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1754.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1754.yaml new file mode 100644 index 000000000..a757f7376 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1754.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1754.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1754 Stereo DAC + +description: + The PCM1754 is a simple stereo DAC that is controlled via hardware gpios. + +maintainers: + - Stefan Kerkmann <s.kerkmann@pengutronix.de> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,pcm1754 + + vcc-supply: true + + '#sound-dai-cells': + const: 0 + + format-gpios: + maxItems: 1 + description: + GPIO used to select the PCM format + + mute-gpios: + maxItems: 1 + description: + GPIO used to mute all outputs + +required: + - compatible + - '#sound-dai-cells' + - vcc-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "ti,pcm1754"; + #sound-dai-cells = <0>; + + vcc-supply = <&vcc_reg>; + mute-gpios = <&gpio 0 GPIO_ACTIVE_HIGH>; + format-gpios = <&gpio 1 GPIO_ACTIVE_HIGH>; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1862.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1862.yaml new file mode 100644 index 000000000..0f0e254a2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1862.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1862.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM186x Universal Audio ADC + +maintainers: + - Ranganath V N <vnranganath.20@gmail.com> + +description: | + The Texas Instruments PCM186x family are multi-channel audio ADCs + that support both I2C and SPI control interfaces, selected by + pin strapping. These devices include on-chip programmable gain + amplifiers and support differential or single-ended analog inputs. + + CODEC input pins: + * VINL1 + * VINR1 + * VINL2 + * VINR2 + * VINL3 + * VINR3 + * VINL4 + * VINR4 + + The pins can be used in referring sound node's audio-routing property. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,pcm1862 + - ti,pcm1863 + - ti,pcm1864 + - ti,pcm1865 + + reg: + maxItems: 1 + + avdd-supply: true + + dvdd-supply: true + + iovdd-supply: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - avdd-supply + - dvdd-supply + - iovdd-supply + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@4a { + compatible = "ti,pcm1865"; + reg = <0x4a>; + + avdd-supply = <®_3v3_analog>; + dvdd-supply = <®_3v3>; + iovdd-supply = <®_1v8>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml b/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml new file mode 100644 index 000000000..0b4f00398 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm3168a.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM3168A Audio Codec + +maintainers: + - Damien Horsley <Damien.Horsley@imgtec.com> + - Geert Uytterhoeven <geert+renesas@glider.be> + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +description: + The Texas Instruments PCM3168A is a 24-bit Multi-channel Audio CODEC with + 96/192kHz sampling rate, supporting both SPI and I2C bus access. + +properties: + compatible: + const: ti,pcm3168a + + reg: + maxItems: 1 + + clocks: + items: + - description: System clock input + + clock-names: + items: + - const: scki + + reset-gpios: + items: + - description: | + GPIO line connected to the active-low RST pin of the codec. + RST = low: device power-down + RST = high: device is enabled + + "#sound-dai-cells": + enum: [0, 1] + + VDD1-supply: + description: Digital power supply regulator 1 (+3.3V) + + VDD2-supply: + description: Digital power supply regulator 2 (+3.3V) + + VCCAD1-supply: + description: ADC power supply regulator 1 (+5V) + + VCCAD2-supply: + description: ADC power supply regulator 2 (+5V) + + VCCDA1-supply: + description: DAC power supply regulator 1 (+5V) + + VCCDA2-supply: + description: DAC power supply regulator 2 (+5V) + + ports: + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + properties: + port@0: + $ref: audio-graph-port.yaml# + description: Audio input port. + + port@1: + $ref: audio-graph-port.yaml# + description: Audio output port. + +required: + - compatible + - reg + - clocks + - clock-names + - VDD1-supply + - VDD2-supply + - VCCAD1-supply + - VCCAD2-supply + - VCCDA1-supply + - VCCDA2-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm3168a: audio-codec@44 { + compatible = "ti,pcm3168a"; + reg = <0x44>; + reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>; + clocks = <&clk_core 42>; + clock-names = "scki"; + VDD1-supply = <&supply3v3>; + VDD2-supply = <&supply3v3>; + VCCAD1-supply = <&supply5v0>; + VCCAD2-supply = <&supply5v0>; + VCCDA1-supply = <&supply5v0>; + VCCDA2-supply = <&supply5v0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml b/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml new file mode 100644 index 000000000..21ea9ff5a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm512x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: PCM512x and TAS575x audio CODECs/amplifiers + +maintainers: + - Animesh Agarwal <animeshagarwal28@gmail.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,pcm5121 + - ti,pcm5122 + - ti,pcm5141 + - ti,pcm5142 + - ti,pcm5242 + - ti,tas5754 + - ti,tas5756 + + reg: + maxItems: 1 + + AVDD-supply: true + + DVDD-supply: true + + CPVDD-supply: true + + clocks: + maxItems: 1 + description: 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. + + '#sound-dai-cells': + const: 0 + + pll-in: + description: GPIO pin used to connect the pll using <1> through <6>. The + device will be configured for clock input on the given pll-in pin. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 6 + + pll-out: + description: GPIO pin used to connect the pll using <1> through <6>. The + device will be configured for PLL output on the given pll-out pin. An + external connection from the pll-out pin to the SCLK pin is assumed. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 6 + +required: + - compatible + - reg + - AVDD-supply + - DVDD-supply + - CPVDD-supply + +if: + properties: + compatible: + contains: + enum: + - ti,tas5754 + - ti,tas5756 + +then: + properties: + pll-in: + maximum: 3 + + pll-out: + maximum: 3 + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@4c { + compatible = "ti,pcm5142"; + reg = <0x4c>; + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + #sound-dai-cells = <0>; + clocks = <&sck>; + pll-in = <3>; + pll-out = <6>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml new file mode 100644 index 000000000..d89b4255b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml @@ -0,0 +1,179 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2024 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm6240.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM6240 Family Audio ADC/DAC + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + The PCM6240 Family is a big family of Audio ADC/DAC for + different Specifications, range from Personal Electric + to Automotive Electric, even some professional fields. + + Specifications about the audio chip can be found at: + https://www.ti.com/lit/gpn/tlv320adc3120 + https://www.ti.com/lit/gpn/tlv320adc5120 + https://www.ti.com/lit/gpn/tlv320adc6120 + https://www.ti.com/lit/gpn/dix4192 + https://www.ti.com/lit/gpn/pcm1690 + https://www.ti.com/lit/gpn/pcm3120-q1 + https://www.ti.com/lit/gpn/pcm3140-q1 + https://www.ti.com/lit/gpn/pcm5120-q1 + https://www.ti.com/lit/gpn/pcm6120-q1 + https://www.ti.com/lit/gpn/pcm6260-q1 + https://www.ti.com/lit/gpn/pcm9211 + https://www.ti.com/lit/gpn/pcmd3140 + https://www.ti.com/lit/gpn/pcmd3180 + https://www.ti.com/lit/gpn/taa5212 + https://www.ti.com/lit/gpn/tad5212 + +properties: + compatible: + description: | + ti,adc3120: Stereo-channel, 768-kHz, Burr-Brownâ„¢ audio analog-to- + digital converter (ADC) with 106-dB SNR. + + ti,adc5120: 2-Channel, 768-kHz, Burr-Brownâ„¢ Audio ADC with 120-dB SNR. + + ti,adc6120: Stereo-channel, 768-kHz, Burr-Brownâ„¢ audio analog-to- + digital converter (ADC) with 123-dB SNR. + + ti,dix4192: 216-kHz digital audio converter with Quad-Channel In + and One-Channel Out. + + ti,pcm1690: Automotive Catalog 113dB SNR 8-Channel Audio DAC with + Differential Outputs. + + ti,pcm3120: Automotive, stereo, 106-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm3140: Automotive, Quad-Channel, 768-kHz, Burr-Brownâ„¢ Audio ADC + with 106-dB SNR. + + ti,pcm5120: Automotive, stereo, 120-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm5140: Automotive, Quad-Channel, 768-kHz, Burr-Brownâ„¢ Audio ADC + with 120-dB SNR. + + ti,pcm6120: Automotive, stereo, 123-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm6140: Automotive, Quad-Channel, 768-kHz, Burr-Brownâ„¢ Audio ADC + with 123-dB SNR. + + ti,pcm6240: Automotive 4-ch audio ADC with integrated programmable mic + bias, boost and input diagnostics. + + ti,pcm6260: Automotive 6-ch audio ADC with integrated programmable mic + bias, boost and input diagnostics. + + ti,pcm9211: 216-kHz digital audio converter With Stereo ADC and + Routing. + + ti,pcmd3140: Four-channel PDM-input to TDM or I2S output converter. + + ti,pcmd3180: Eight-channel pulse-density-modulation input to TDM or + I2S output converter. + + ti,taa5212: Low-power high-performance stereo audio ADC with 118-dB + dynamic range. + + ti,tad5212: Low-power stereo audio DAC with 120-dB dynamic range. + oneOf: + - items: + - enum: + - ti,adc3120 + - ti,adc5120 + - ti,pcm3120 + - ti,pcm5120 + - ti,pcm6120 + - const: ti,adc6120 + - items: + - enum: + - ti,pcmd512x + - ti,pcm9211 + - ti,taa5212 + - ti,tad5212 + - const: ti,adc6120 + - items: + - enum: + - ti,pcm3140 + - ti,pcm5140 + - ti,dix4192 + - ti,pcm6140 + - ti,pcm6260 + - const: ti,pcm6240 + - items: + - enum: + - ti,pcmd3140 + - ti,pcmd3180 + - ti,pcm1690 + - ti,taa5412 + - ti,tad5412 + - const: ti,pcm6240 + - enum: + - ti,adc6120 + - ti,pcm6240 + + reg: + description: + I2C address, in multiple pcmdevices case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + minItems: 1 + maxItems: 4 + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + description: + Invalid only for ti,pcm1690 because of no INT pin. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,pcm1690 + then: + properties: + interrupts: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + /* example for two devices with interrupt support */ + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@48 { + compatible = "ti,pcm6240"; + reg = <0x48>, /* primary-device */ + <0x4b>; /* secondary-device */ + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + interrupt-parent = <&gpio1>; + interrupts = <15>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000..27230c682 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax@flatmax.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas2552.yaml b/Documentation/devicetree/bindings/sound/ti,tas2552.yaml new file mode 100644 index 000000000..85e3ebd2a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2552.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2552.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2552 Codec + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + - Kevin Lu <kevin-lu@ti.com> + - Baojun Xu <baojun.xu@ti.com> + +description: > + The 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. + +properties: + compatible: + const: ti,tas2552 + + reg: + maxItems: 1 + + vbat-supply: true + iovdd-supply: true + avdd-supply: true + + enable-gpio: + maxItems: 1 + description: gpio pin to enable/disable the device + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - vbat-supply + - iovdd-supply + - avdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + #sound-dai-cells = <0>; + vbat-supply = <®_vbat>; + iovdd-supply = <®_iovdd>; + avdd-supply = <®_avdd>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tas2562.yaml b/Documentation/devicetree/bindings/sound/ti,tas2562.yaml new file mode 100644 index 000000000..3763ca16b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2562.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2562.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Andrew Davis <afd@ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2562 + https://www.ti.com/lit/gpn/tas2564 + https://www.ti.com/lit/gpn/tas2110 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2564 + - ti,tas2110 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + ti,imon-slot-no = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tas2770.yaml b/Documentation/devicetree/bindings/sound/ti,tas2770.yaml new file mode 100644 index 000000000..8d49fbcf0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2770.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,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 + - ti,tas5770l # Apple variant + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x41 to 0x48. + + 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. + + ti,asi-format: + deprecated: true + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@41 { + compatible = "ti,tas2770"; + reg = <0x41>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + shutdown-gpios = <&gpio1 14 GPIO_ACTIVE_HIGH>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000..aa5a317b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,309 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2026 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2563/TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + The TAS2118/TAS2X20 is mono, digital input Class-D audio + amplifier optimized for efficiently driving high peak power into + small loudspeakers. + The TAS257x is 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. + The TAS2563/TAS2781 is a mono, digital input Class-D audio + amplifier optimized for efficiently driving high peak power into + small loudspeakers. An integrated on-chip DSP supports Texas + Instruments Smart Amp speaker protection algorithm. The + integrated speaker voltage and current sense provides for real time + monitoring of loudspeaker behavior. + The TAS5802/TAS5815/TAS5822/TAS5825/TAS5827/TAS5828 is a stereo, + 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. + + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2120 + https://www.ti.com/lit/gpn/tas2320 + https://www.ti.com/lit/gpn/tas2563 + https://www.ti.com/lit/gpn/tas2572 + https://www.ti.com/lit/gpn/tas2574 + https://www.ti.com/lit/gpn/tas2781 + https://www.ti.com/lit/gpn/tas5806m + https://www.ti.com/lit/gpn/tas5806md + https://www.ti.com/lit/gpn/tas5815 + https://www.ti.com/lit/gpn/tas5822m + https://www.ti.com/lit/gpn/tas5825m + https://www.ti.com/lit/gpn/tas5827 + https://www.ti.com/lit/gpn/tas5828m + https://www.ti.com/lit/gpn/tas5830 + +properties: + compatible: + description: | + ti,tas2020: 3.2-W Mono Digital Input Class-D Speaker Amp with 5.5V PVDD + Support. + + ti,tas2118: 5-W Mono Digital Input Class-D Speaker Amp with Integrated + 8.4-V Class-H Boost. + + ti,tas2120: 8.2-W Mono Digital Input Class-D Speaker Amp with + Integrated 14.75V Class-H Boost. + + ti,tas2320: 15-W Mono Digital Input Class-D Speaker Amp with 15V Support. + + ti,tas2563: 6.1-W Boosted Class-D Audio Amplifier With Integrated + DSP and IV Sense, 16/20/24/32bit stereo I2S or multichannel TDM. + + ti,tas2568: 5.3-W Digital Input Smart Amp with I/V Sense and Integrated + 10.75-V Class-H Boost + + ti,tas2570: 5.8-W Digital Input smart amp with I/V sense and integrated + 11-V Class-H Boost + + ti,tas2572: 6.6-W Digital Input smart amp with I/V sense and integrated + 13-V Class-H Boost + + ti,tas2573: Boosted Class-D Audio Amplifier with Integrated DSP and IV + Sense + + ti,tas2574: 8.5-W Digital Input smart amp with I/V sense and integrated + 15-V Class-H Boost + + ti,tas2781: 24-V Class-D Amplifier with Real Time Integrated Speaker + Protection and Audio Processing, 16/20/24/32bit stereo I2S or + multichannel TDM. + + ti,tas5802: 22-W, Inductor-Less, Digital Input, Closed-Loop Class-D + Audio Amplifier with 96-Khz Extended Processing and Low Idle Power + Dissipation. + + ti,tas5806m: 23-W, Inductor-Less, Digital Input, Stereo, Closed-Loop + Class-D Audio Amplifier with Enhanced Processing and Low Power + Dissipation. + + ti,tas5806md: 23-W, Inductor-Less, Digital Input, Stereo, Closed-Loop + Class-D Audio Amplifier with Enhanced Processing and DirectPath(TM) + HP Driver + + ti,tas5815: 30-W, Digital Input, Stereo, Closed-loop Class-D Audio + Amplifier with 96 kHz Enhanced Processing + + ti,tas5822: 35-W, Digital Input, Stereo, Closed-Loop Class-D Audio + Amplifier with 96 kHz Enhanced Processing + + ti,tas5825: 38-W Stereo, Inductor-Less, Digital Input, Closed-Loop 4.5V + to 26.4V Class-D Audio Amplifier with 192-kHz Extended Audio Processing. + + ti,tas5827: 47-W Stereo, Digital Input, High Efficiency Closed-Loop + Class-D Amplifier with Class-H Algorithm + + ti,tas5828: 50-W Stereo, Digital Input, High Efficiency Closed-Loop + Class-D Amplifier with Hybrid-Pro Algorithm + + ti,tas5830: 65-W Stereo, Digital Input, High Efficiency Closed-Loop + Class-D Amplifier with Class-H Algorithm + + ti,tas5832: 81-W Stereo, Digital Input, High Efficiency Closed-Loop + Class-D Amplifier with Class-H Algorithm + oneOf: + - items: + - enum: + - ti,tas2020 + - ti,tas2118 + - ti,tas2120 + - ti,tas2320 + - ti,tas2563 + - ti,tas2568 + - ti,tas2570 + - ti,tas2572 + - ti,tas2573 + - ti,tas2574 + - ti,tas5802 + - ti,tas5806m + - ti,tas5806md + - ti,tas5815 + - ti,tas5822 + - ti,tas5825 + - ti,tas5827 + - ti,tas5828 + - ti,tas5830 + - ti,tas5832 + - const: ti,tas2781 + - enum: + - ti,tas2781 + + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 8 + minItems: 1 + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,tas2020 + - ti,tas2118 + - ti,tas2120 + - ti,tas2320 + - ti,tas2568 + - ti,tas2574 + then: + properties: + reg: + maxItems: 4 + items: + minimum: 0x48 + maximum: 0x4b + + - if: + properties: + compatible: + contains: + enum: + - ti,tas2563 + - ti,tas2573 + - ti,tas5825 + then: + properties: + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 4 + minItems: 1 + items: + minimum: 0x4c + maximum: 0x4f + + - if: + properties: + compatible: + contains: + enum: + - ti,tas2570 + - ti,tas2572 + then: + properties: + reg: + maxItems: 4 + items: + minimum: 0x48 + maximum: 0x4b + + - if: + properties: + compatible: + contains: + enum: + - ti,tas2781 + then: + properties: + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 8 + minItems: 1 + items: + minimum: 0x38 + maximum: 0x3f + + - if: + properties: + compatible: + contains: + enum: + - ti,tas5802 + - ti,tas5815 + then: + properties: + reg: + maxItems: 4 + items: + minimum: 0x54 + maximum: 0x57 + + - if: + properties: + compatible: + contains: + enum: + - ti,tas5806m + - ti,tas5806md + - ti,tas5822 + then: + properties: + reg: + maxItems: 4 + items: + minimum: 0x2c + maximum: 0x2f + + - if: + properties: + compatible: + contains: + enum: + - ti,tas5827 + - ti,tas5828 + - ti,tas5830 + - ti,tas5832 + then: + properties: + reg: + maxItems: 6 + items: + minimum: 0x60 + maximum: 0x65 + +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>; + + audio-codec@38 { + compatible = "ti,tas2781"; + reg = <0x38>, /* Audio slot 0 */ + <0x3a>, /* Audio slot 1 */ + <0x39>, /* Audio slot 2 */ + <0x3b>; /* Audio slot 3 */ + + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + interrupt-parent = <&gpio1>; + interrupts = <15>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas27xx.yaml b/Documentation/devicetree/bindings/sound/ti,tas27xx.yaml new file mode 100644 index 000000000..fcaae848e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas27xx.yaml @@ -0,0 +1,85 @@ +# 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/ti,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 + - ti,sn012776 # Apple variant of TAS2764 + + 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@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <0>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + shutdown-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 000000000..234dad296 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,48 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + + - ti,mid-z-channel-X: Boolean properties, X being a number from 1 to 6. + If given, channel X will start with the Mid-Z start + sequence, otherwise the default Low-Z scheme is used. + + The correct configuration depends on how the power + stages connected to the PWM output pins work. Not all + power stages are compatible to Mid-Z - please refer + to the datasheets for more details. + + Most systems should not set any of these properties. + + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tas57xx.yaml b/Documentation/devicetree/bindings/sound/ti,tas57xx.yaml new file mode 100644 index 000000000..0b013a34e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas57xx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas57xx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers + +maintainers: + - Neil Armstrong <neil.armstrong@linaro.org> + +properties: + compatible: + enum: + - ti,tas5707 + - ti,tas5711 + - ti,tas5717 + - ti,tas5719 + - ti,tas5721 + - ti,tas5733 + - ti,tas5753 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: GPIO for the active low reset line + + pdn-gpios: + maxItems: 1 + description: GPIO for the active low powerdown line + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + AVDD-supply: true + DVDD-supply: true + HPVDD-supply: true + PVDD_AB-supply: true + PVDD_CD-supply: true + PVDD_A-supply: true + PVDD_B-supply: true + PVDD_C-supply: true + PVDD_D-supply: true + DRVDD-supply: true + PVDD-supply: true + + '#sound-dai-cells': + const: 0 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - '#sound-dai-cells' + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,tas5717 + - ti,tas5719 + then: + properties: + PVDD_A-supply: false + PVDD_B-supply: false + PVDD_C-supply: false + PVDD_D-supply: false + DRVDD-supply: false + PVDD-supply: false + + - if: + properties: + compatible: + contains: + enum: + - ti,tas5711 + then: + properties: + HPVDD-supply: false + PVDD_AB-supply: false + PVDD_CD-supply: false + DRVDD-supply: false + PVDD-supply: false + + - if: + properties: + compatible: + contains: + enum: + - ti,tas5721 + - ti,tas5753 + then: + properties: + HPVDD-supply: false + PVDD_AB-supply: false + PVDD_CD-supply: false + PVDD_A-supply: false + PVDD_B-supply: false + PVDD_C-supply: false + PVDD_D-supply: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@2a { + compatible = "ti,tas5717"; + reg = <0x2a>; + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + pdn-gpios = <&gpio1 15 GPIO_ACTIVE_HIGH>; + AVDD-supply = <&avdd_supply>; + DVDD-supply = <&dvdd_supply>; + HPVDD-supply = <&hpvdd_supply>; + PVDD_AB-supply = <&pvdd_ab_supply>; + PVDD_CD-supply = <&pvdd_cd_supply>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas5805m.yaml b/Documentation/devicetree/bindings/sound/ti,tas5805m.yaml new file mode 100644 index 000000000..c2c2835a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5805m.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas5805m.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: TAS5805M audio amplifier + +maintainers: + - Daniel Beer <daniel.beer@igorinstitute.com> + +description: | + The TAS5805M is a class D audio amplifier with a built-in DSP. + +properties: + compatible: + enum: + - ti,tas5805m + + reg: + maxItems: 1 + description: | + I2C address of the amplifier. See the datasheet for possible values. + + pvdd-supply: + description: | + Regulator for audio power supply (PVDD in the datasheet). + + pdn-gpios: + description: | + Power-down control GPIO (PDN pin in the datasheet). + + ti,dsp-config-name: + description: | + The name of the DSP configuration that should be loaded for this + instance. Configuration blobs are sequences of register writes + generated from TI's PPC3 tool. + $ref: /schemas/types.yaml#/definitions/string + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + tas5805m: tas5805m@2c { + reg = <0x2c>; + compatible = "ti,tas5805m"; + + pvdd-supply = <&audiopwr>; + pdn-gpios = <&tlmm 160 0>; + + ti,dsp-config-name = "mono_pbtl_48khz"; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt new file mode 100644 index 000000000..00940c489 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt @@ -0,0 +1,22 @@ +Texas Instruments TAS6424 Quad-Channel Audio amplifier + +The TAS6424 serial control bus communicates through I2C protocols. + +Required properties: + - compatible: "ti,tas6424" - TAS6424 + - reg: I2C slave address + - sound-dai-cells: must be equal to 0 + - standby-gpios: GPIO used to shut the TAS6424 down. + - mute-gpios: GPIO used to mute all the outputs + +Example: + +tas6424: tas6424@6a { + compatible = "ti,tas6424"; + reg = <0x6a>; + + #sound-dai-cells = <0>; +}; + +For more product information please see the link below: +https://www.ti.com/product/TAS6424-Q1 diff --git a/Documentation/devicetree/bindings/sound/ti,tas67524.yaml b/Documentation/devicetree/bindings/sound/ti,tas67524.yaml new file mode 100644 index 000000000..812a4d39e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas67524.yaml @@ -0,0 +1,280 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas67524.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS67524 Audio Amplifier + +maintainers: + - Sen Wang <sen@ti.com> + +description: + The TAS67524 is a four-channel, digital-input, automotive + Class-D audio amplifier with load diagnostics and an integrated + DSP for audio processing. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + oneOf: + - items: + - enum: + - ti,tas6754 + - const: ti,tas67524 + - const: ti,tas67524 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + description: | + The device exposes three DAIs, selected by index. + 0 - Standard Audio Path (Playback) + 1 - Low-Latency Playback Path (Playback) + 2 - Sensory Feedback (Capture - Vpredict and Isense) + By default, all four channels of each DAI are active. + + interrupts: + maxItems: 1 + description: + Active-low falling-edge interrupt from the FAULT pin. When provided, + the driver uses IRQ-driven fault reporting instead of polling. + + powerdown-gpios: + maxItems: 1 + description: + GPIO connected to the PD pin, active low. Controls the internal + digital circuitry power state. When asserted the device enters + full power-down mode and all register state is lost. Can be omitted if + PD pin is hardwired or externally controlled. + + standby-gpios: + maxItems: 1 + description: + GPIO connected to the STBY pin, active low. Controls the analog + power stage. When asserted the device enters Deep Sleep mode but + remains I2C-accessible with registers retained. Can be omitted if + STBY pin is tied to PD or hardwired. + + dvdd-supply: + description: + Digital logic supply (1.62 V to 3.6 V). All three supply rails must + be within their recommended operating ranges before the PD pin is + released. + + pvdd-supply: + description: + Output FET power supply (4.5 V to 19 V). All three supply rails must + be within their recommended operating ranges before the PD pin is + released. + + vbat-supply: + description: + Battery supply for the Class-D output stage (4.5 V to 19 V). Optional + when PVDD and VBAT are connected to the same supply rail. When absent, + VBAT is assumed hardwired to PVDD. + + ti,fast-boot: + type: boolean + description: + Skip DC load diagnostic sweep at power-on to reduce boot latency. + Automatic diagnostics after fault conditions remain enabled. Hardware + overcurrent protection is always active. + + ti,audio-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + TDM slot offset for the standard audio playback path via SDIN1. A value + of 4 maps to slot 4. If omitted, slot assignment is derived from the + tx_mask provided via set_tdm_slot(). Without either property, no slot + mapping is configured. + + ti,llp-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + TDM slot offset for the low-latency playback path via SDIN1. If omitted, + slot assignment is derived from the tx_mask provided via set_tdm_slot(). + Without either property, no slot mapping is configured. Disabled outside + of LLP mode, and only relevant for TDM formats. + + ti,vpredict-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + In TDM mode, enables Vpredict output and assigns its starting slot; + four consecutive slots carry Vpredict Ch1-4 on SDOUT1. May coexist + with ti,isense-slot-no using separate non-overlapping slots. + + In I2S mode, enables Vpredict output on SDOUT1 (Ch1/Ch2) and SDOUT2 + (Ch3/Ch4). The slot value is unused. Requires a GPIO configured as + sdout2 for Ch3/Ch4; without it only Ch1/Ch2 are output. Mutually + exclusive with ti,isense-slot-no; if both are set, Vpredict takes + priority. + + Irrelevant in Left-J and Right-J modes. + + ti,isense-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + In TDM mode, enables Isense output and assigns its starting slot; + four consecutive slots carry Isense Ch1-4 on SDOUT1. May coexist + with ti,vpredict-slot-no using separate non-overlapping slots. + + In I2S mode, enables Isense output on SDOUT1 (Ch1/Ch2) and SDOUT2 + (Ch3/Ch4). The slot value is unused. Requires a GPIO configured as + SDOUT2 for Ch3/Ch4; without it only Ch1/Ch2 are output. Mutually + exclusive with ti,vpredict-slot-no; Vpredict takes priority if both + are set. + + Irrelevant in Left-J and Right-J modes. + + ti,gpio1-function: + $ref: /schemas/types.yaml#/definitions/string + description: + Function for the GPIO_1 pin. When omitted, GPIO_1 remains in its + power-on default state. + enum: + - low # Output: driven low + - auto-mute # Output: high when all channels are auto-muted + - auto-mute-ch4 # Output: high when channel 4 is auto-muted + - auto-mute-ch3 # Output: high when channel 3 is auto-muted + - auto-mute-ch2 # Output: high when channel 2 is auto-muted + - auto-mute-ch1 # Output: high when channel 1 is auto-muted + - sdout2 # Output: Routes secondary serial data output 2 + - sdout1 # Output: Re-routes secondary serial data output 1 + - warn # Output: warning signal (OTW, CBC) + - fault # Output: fault signal (OTSD, OC, DC) + - clock-sync # Output: clock synchronisation + - invalid-clock # Output: high when clock is invalid + - high # Output: driven high + - mute # Input: external mute control + - phase-sync # Input: phase synchronisation + - sdin2 # Input: secondary SDIN2 for I2S/LJ/RJ ch3/ch4 + - deep-sleep # Input: asserted transitions device to Deep Sleep + - hiz # Input: asserted transitions device to Hi-Z + - play # Input: asserted transitions device to Play + - sleep # Input: asserted transitions device to Sleep + + ti,gpio2-function: + $ref: /schemas/types.yaml#/definitions/string + description: + Function for the GPIO_2 pin. When omitted, GPIO_2 remains in its + power-on default state. + enum: + - low # Output: driven low + - auto-mute # Output: high when all channels are auto-muted + - auto-mute-ch4 # Output: high when channel 4 is auto-muted + - auto-mute-ch3 # Output: high when channel 3 is auto-muted + - auto-mute-ch2 # Output: high when channel 2 is auto-muted + - auto-mute-ch1 # Output: high when channel 1 is auto-muted + - sdout2 # Output: Routes secondary serial data output 2 + - sdout1 # Output: Re-routes secondary serial data output 1 + - warn # Output: warning signal (OTW, CBC) + - fault # Output: fault signal (OTSD, OC, DC) + - clock-sync # Output: clock synchronisation + - invalid-clock # Output: high when clock is invalid + - high # Output: driven high + - mute # Input: external mute control + - phase-sync # Input: phase synchronisation + - sdin2 # Input: secondary SDIN2 for I2S/LJ/RJ ch3/ch4 + - deep-sleep # Input: asserted transitions device to Deep Sleep + - hiz # Input: asserted transitions device to Hi-Z + - play # Input: asserted transitions device to Play + - sleep # Input: asserted transitions device to Sleep + + ports: + $ref: /schemas/graph.yaml#/properties/ports + properties: + port@0: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Standard audio playback port (DAI 0). + + port@1: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Low-latency playback port (LLP) (DAI 1). + + port@2: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + description: Sensory feedback capture port (DAI 2). + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - '#sound-dai-cells' + - dvdd-supply + - pvdd-supply + +anyOf: + - required: [powerdown-gpios] + - required: [standby-gpios] + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@70 { + compatible = "ti,tas67524"; + reg = <0x70>; + #sound-dai-cells = <1>; + sound-name-prefix = "TAS0"; + + standby-gpios = <&main_gpio0 33 GPIO_ACTIVE_LOW>; + + dvdd-supply = <&dvdd_1v8>; + pvdd-supply = <&pvdd_12v>; + vbat-supply = <&vbat_12v>; + + ti,audio-slot-no = <0>; + ti,llp-slot-no = <4>; + ti,vpredict-slot-no = <0>; + ti,isense-slot-no = <4>; + + ti,gpio2-function = "warn"; + + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0>; + + tas0_audio_ep: endpoint { + dai-format = "dsp_b"; + remote-endpoint = <&be_tas0_audio_ep>; + }; + }; + + port@1 { + reg = <1>; + + tas0_anc_ep: endpoint { + remote-endpoint = <&be_tas0_anc_ep>; + }; + }; + + port@2 { + reg = <2>; + + tas0_fb_ep: endpoint { + remote-endpoint = <&be_tas0_fb_ep>; + }; + }; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000..66b766562 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,164 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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 selected, the pin may be controlled via the + GPIO framework, as pin number 0 on the device. + + 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 selected, the pin may be controlled via the + GPIO framework, as pin number 1 on the device. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-gpo: + type: boolean + description: | + When set, the MICBIAS1 pin may be controlled via the GPIO framework, + as pin number 3 on the device. + + In this mode, when the pin is activated, it will be set to the voltage + specified by the ti,micbias1-vg property. When deactivated, the pin will + float. + + ti,micbias2-gpo: + type: boolean + description: | + When set, the MICBIAS2 pin may be controlled via the GPIO framework, + as pin number 4 on the device. + + In this mode, when the pin is activated, it will be set to the voltage + specified by the ti,micbias2-vg property. When deactivated, the pin will + float. + + 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: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +dependencies: + ti,micbias1-gpo: ['ti,micbias1-vg'] + ti,micbias2-gpo: ['ti,micbias2-vg'] + +required: + - compatible + - reg + - clocks + +unevaluatedProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: audio-codec@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml new file mode 100644 index 000000000..a93de2deb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml @@ -0,0 +1,212 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,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: | + External supply of 1.8V. If not defined then the internal regulator is + enabled instead. + + avdd-supply: true + iovdd-supply: true + + 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 + + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + + ti,asi-tx-drive: + type: boolean + description: | + When set the device will set the Tx ASI output to a Hi-Z state for unused + data cycles. Default is to drive the output low on unused ASI cycles. + +patternProperties: + '^ti,gpo-config-[1-4]$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + Defines the configuration and output driver for the general purpose + output pins (GPO). These values are pairs, the first value is for the + configuration type and the second value is for the output drive type. + The array is defined as <GPO_CFG GPO_DRV> + + GPO output configuration can be one of the following: + + 0 - (default) disabled + 1 - GPOX is configured as a general-purpose output (GPO) + 2 - GPOX is configured as a device interrupt output (IRQ) + 3 - GPOX is configured as a secondary ASI output (SDOUT2) + 4 - GPOX is configured as a PDM clock output (PDMCLK) + + GPO output drive configuration for the GPO pins can be one of the following: + + 0d - (default) Hi-Z output + 1d - Drive active low and active high + 2d - Drive active low and weak high + 3d - Drive active low and Hi-Z + 4d - Drive weak low and active high + 5d - Drive Hi-Z and active high + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tlv320adc5140"; + reg = <0x4c>; + ti,mic-bias-source = <6>; + ti,pdm-edge-select = <0 1 0 1>; + ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; + ti,gpo-config-1 = <0 0>; + ti,gpo-config-2 = <0 0>; + reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml new file mode 100644 index 000000000..4783e6dbb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic32x4.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC32x4 Stereo Audio codec + +maintainers: + - Alexander Stein <alexander.stein@ew.tq-group.com> + +description: | + The TLV320AIC32x4 audio codec can be accessed using I2C or SPI + +properties: + compatible: + enum: + - ti,tas2505 + - ti,tlv320aic32x4 + - ti,tlv320aic32x6 + + reg: + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + av-supply: + description: Analog core power supply + + dv-supply: + description: Digital core power supply + + iov-supply: + description: Digital IO power supply + + ldoin-supply: + description: LDO power supply + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + aic32x4-gpio-func: + description: | + GPIO function configuration for pins MFP1-MFP5. + Types are defined in include/sound/tlv320aic32x4.h + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + - clocks + - clock-names + - iov-supply + +allOf: + - $ref: dai-common.yaml# + - if: + not: + required: + - ldoin-supply + then: + required: + - av-supply + - dv-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; + iov-supply = <®_3v3>; + ldoin-supply = <®_3v3>; + clocks = <&clks 201>; + clock-names = "mclk"; + aic32x4-gpio-func = < + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml new file mode 100644 index 000000000..50088698a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -0,0 +1,167 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +# Copyright (C) 2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$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,tlv320aic23 + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + gpio-reset: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + maxItems: 1 + description: + Deprecated, please use reset-gpios instead. + deprecated: true + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + + clocks: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: audio-codec@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: audio-codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml new file mode 100644 index 000000000..10299064c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml @@ -0,0 +1,126 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320dac3100.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tlv320aic31xx Codec module + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + 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. + +properties: + compatible: + enum: + - 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: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + HPVDD-supply: true + + SPRVDD-supply: true + + SPLVDD-supply: true + + AVDD-supply: true + + IOVDD-supply: true + + DVDD-supply: true + + reset-gpios: + description: GPIO specification for the active low RESET input. + + ai31xx-micbias-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: [1, 2, 3] + description: | + 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 + + ai31xx-ocmv: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + + gpio-reset: + description: gpio pin number used for codec reset + deprecated: true + +required: + - compatible + - reg + - HPVDD-supply + - SPRVDD-supply + - SPLVDD-supply + - AVDD-supply + - IOVDD-supply + - DVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320aic31xx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + sound@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_2_0V>; + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 000000000..a42bf9bde --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre@kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml b/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml new file mode 100644 index 000000000..785930658 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,ts3a227e.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TS3A227E + Autonomous Audio Accessory Detection and Configuration Switch + +maintainers: + - Dylan Reid <dgreid@chromium.org> + +description: | + The TS3A227E detect headsets of 3-ring and 4-ring standards and + switches automatically to route the microphone correctly. It also + handles key press detection in accordance with the Android audio + headset specification v1.0. + +properties: + compatible: + enum: + - ti,ts3a227e + + reg: + const: 0x3b + + interrupts: + maxItems: 1 + + ti,micbias: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Intended MICBIAS voltage (datasheet section 9.6.7). + enum: + - 0 # 2.1 V + - 1 # 2.2 V + - 2 # 2.3 V + - 3 # 2.4 V + - 4 # 2.5 V + - 5 # 2.6 V + - 6 # 2.7 V + - 7 # 2.8 V + default: 1 + + ti,debounce-release-ms: + description: key release debounce time in ms (datasheet section 9.6.7). + enum: + - 0 + - 20 + default: 20 + + ti,debounce-press-ms: + description: key press debounce time in ms (datasheet section 9.6.7). + enum: + - 2 + - 40 + - 80 + - 120 + default: 80 + + ti,debounce-insertion-ms: + description: headset insertion debounce time in ms (datasheet section 9.6.5). + enum: + - 2 + - 30 + - 60 + - 90 + - 120 + - 150 + - 1000 + - 2000 + default: 90 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: audio-controller@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio1>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ti,twl4030-audio.yaml b/Documentation/devicetree/bindings/sound/ti,twl4030-audio.yaml new file mode 100644 index 000000000..c9c3f7513 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,twl4030-audio.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,twl4030-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TWL4030-family Audio Module + +maintainers: + - Peter Ujfalusi <peter.ujfalusi@gmail.com> + +description: + The audio module within the TWL4030-family of companion chips consists + of an audio codec and a vibra driver. This binding describes the parent + node for these functions. + +properties: + compatible: + const: ti,twl4030-audio + + codec: + type: object + description: Node containing properties for the audio codec functionality. + + properties: + ti,digimic_delay: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Delay in milliseconds after enabling digital microphones to reduce + artifacts. + + ti,ramp_delay_value: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Headset ramp delay configuration to reduce pop noise. + + ti,hs_extmute: + type: boolean + description: + Enable the use of an external mute for headset pop reduction. + + ti,hs_extmute_gpio: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: + The GPIO specifier for the external mute control. + maxItems: 1 + + ti,offset_cncl_path: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Offset cancellation path selection. Refer to the Technical + Reference Manual for valid values. + + # The 'codec' node itself is optional, but if it exists, it can be empty. + # We don't require any of its sub-properties. + + ti,enable-vibra: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + description: + Enable or disable the vibra functionality. + +additionalProperties: false + +required: + - compatible + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + twl: twl@48 { + reg = <0x48>; + interrupts = <7>; /* SYS_NIRQ cascaded to intc */ + interrupt-parent = <&intc>; + + twl_audio: audio { + compatible = "ti,twl4030-audio"; + + ti,enable-vibra = <1>; + + codec { + ti,ramp_delay_value = <3>; + }; + + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/trivial-codec.yaml b/Documentation/devicetree/bindings/sound/trivial-codec.yaml new file mode 100644 index 000000000..9a35dfb17 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/trivial-codec.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/trivial-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Trivial Audio Codec + +maintainers: + - Rob Herring <robh@kernel.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + # Analog Devices SSM2602 I2S audio CODEC devices + - adi,ssm2602 + - adi,ssm2603 + - adi,ssm2604 + - adi,ssm3515 + # Cirrus Logic CS4265 audio DAC + - cirrus,cs4265 + - cirrus,cs4341a + - cirrus,cs4349 + - dlg,da9055-codec + # Nuvoton Technology Corporation NAU85L40 Audio CODEC + - nuvoton,nau8540 + - nuvoton,nau8810 + - nuvoton,nau8812 + - nuvoton,nau8814 + # NXP TFA9879 class-D audio amplifier + - nxp,tfa9879 + - nxp,uda1342 + - sdw3019f836300 + - ti,pcm1789 + - ti,pcm1792a + - ti,pcm5102a + - wlf,wm8510 + - wlf,wm8523 + - wlf,wm8580 + - wlf,wm8581 + - wlf,wm8711 + - wlf,wm8728 + - wlf,wm8737 + - wlf,wm8750 + - wlf,wm8753 + - wlf,wm8770 + - wlf,wm8776 + - wlf,wm8961 + - wlf,wm8974 + - wlf,wm8987 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8523"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt new file mode 100644 index 000000000..7eea32e9d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -0,0 +1,22 @@ +TSCS42XX Audio CODEC + +Required Properties: + + - compatible : "tempo,tscs42A1" for analog mic + "tempo,tscs42A2" for digital mic + + - reg : <0x71> for analog mic + <0x69> for digital mic + + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + +Example: + +wookie: codec@69 { + compatible = "tempo,tscs42A2"; + reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; +}; diff --git a/Documentation/devicetree/bindings/sound/tscs454.txt b/Documentation/devicetree/bindings/sound/tscs454.txt new file mode 100644 index 000000000..3ba3e2d2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tscs454.txt @@ -0,0 +1,23 @@ +TSCS454 Audio CODEC + +Required Properties: + + - compatible : "tempo,tscs454" + + - reg : <0x69> + + - clock-names: Must one of the following "xtal", "mclk1", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + + Note: If clock is not provided then bit clock is assumed + +Example: + +redwood: codec@69 { + #sound-dai-cells = <1>; + compatible = "tempo,tscs454"; + reg = <0x69>; + clock-names = "mclk1"; + clocks = <&audio_mclk>; +}; diff --git a/Documentation/devicetree/bindings/sound/uda1334.txt b/Documentation/devicetree/bindings/sound/uda1334.txt new file mode 100644 index 000000000..f64071b25 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/uda1334.txt @@ -0,0 +1,17 @@ +UDA1334 audio CODEC + +This device uses simple GPIO pins for controlling codec settings. + +Required properties: + + - compatible : "nxp,uda1334" + - nxp,mute-gpios: a GPIO spec for the MUTE pin. + - nxp,deemph-gpios: a GPIO spec for the De-emphasis pin + +Example: + +uda1334: audio-codec { + compatible = "nxp,uda1334"; + nxp,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + nxp,deemph-gpios = <&gpio3 3 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/devicetree/bindings/sound/ux500-mop500.txt b/Documentation/devicetree/bindings/sound/ux500-mop500.txt new file mode 100644 index 000000000..48e071c96 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-mop500.txt @@ -0,0 +1,39 @@ +* MOP500 Audio Machine Driver + +This node is responsible for linking together all ux500 Audio Driver components. + +Required properties: + - compatible : "stericsson,snd-soc-mop500" + +Non-standard properties: + - stericsson,cpu-dai : Phandle to the CPU-side DAI + - stericsson,audio-codec : Phandle to the Audio CODEC + - stericsson,card-name : Over-ride default card name + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; diff --git a/Documentation/devicetree/bindings/sound/ux500-msp.txt b/Documentation/devicetree/bindings/sound/ux500-msp.txt new file mode 100644 index 000000000..7dd1b9616 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-msp.txt @@ -0,0 +1,42 @@ +* ux500 MSP (CPU-side Digital Audio Interface) + +Required properties: + - compatible :"stericsson,ux500-msp-i2s" + - reg : Physical base address and length of the device's registers. + +Optional properties: + - interrupts : The interrupt output from the device. + - <name>-supply : Phandle to the regulator <name> supply + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,platform-pcm-dma = <&pcm>; + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + pcm: ux500-pcm { + compatible = "stericsson,ux500-pcm"; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; diff --git a/Documentation/devicetree/bindings/sound/widgets.txt b/Documentation/devicetree/bindings/sound/widgets.txt new file mode 100644 index 000000000..b6de5ba3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/widgets.txt @@ -0,0 +1,20 @@ +Widgets: + +This mainly specifies audio off-codec DAPM widgets. + +Each entry is a pair of strings in DT: + + "template-wname", "user-supplied-wname" + +The "template-wname" being the template widget name and currently includes: +"Microphone", "Line", "Headphone" and "Speaker". + +The "user-supplied-wname" being the user specified widget name. + +For instance: + simple-audio-widgets = + "Microphone", "Microphone Jack", + "Line", "Line In Jack", + "Line", "Line Out Jack", + "Headphone", "Headphone Jack", + "Speaker", "Speaker External"; diff --git a/Documentation/devicetree/bindings/sound/wlf,arizona.yaml b/Documentation/devicetree/bindings/sound/wlf,arizona.yaml new file mode 100644 index 000000000..8156f30ea --- /dev/null +++ b/Documentation/devicetree/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. + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 12 + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml new file mode 100644 index 000000000..dffc41d5f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - wlf,mute-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + wm8524: codec { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml new file mode 100644 index 000000000..858c0f689 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson Microelectromics WM8731 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +description: | + Wolfson Microelectronics WM8731 audio CODEC + + Pins on the device (for linking into audio routes): + * LOUT: Left Channel Line Output + * ROUT: Right Channel Line Output + * LHPOUT: Left Channel Headphone Output + * RHPOUT: Right Channel Headphone Output + * LLINEIN: Left Channel Line Input + * RLINEIN: Right Channel Line Input + * MICIN: Microphone Input + +properties: + compatible: + enum: + - wlf,wm8731 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + description: Clock provider for MCLK pin. + maxItems: 1 + + clock-names: + items: + - const: mclk + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + HPVDD-supply: + description: Headphone power supply regulator on the HPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + +required: + - reg + - compatible + - AVDD-supply + - HPVDD-supply + - DBVDD-supply + - DCVDD-supply + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + wm8731_i2c: codec@0 { + compatible = "wlf,wm8731"; + reg = <0>; + spi-max-frequency = <12500000>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; + - | + + i2c { + #address-cells = <1>; + #size-cells = <0>; + wm8731_spi: codec@1b { + compatible = "wlf,wm8731"; + reg = <0x1b>; + + AVDD-supply = <&avdd_reg>; + HPVDD-supply = <&hpvdd_reg>; + DCVDD-supply = <&dcvdd_reg>; + DBVDD-supply = <&dbvdd_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8782.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8782.yaml new file mode 100644 index 000000000..d0bbdc9f9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8782.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8782.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson Microelectromics WM8782 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8782 + + Vdda-supply: + description: Regulator for the analog power supply (2.7V - 5.5V) + + Vdd-supply: + description: Regulator for the digital power supply (2.7V - 3.6V) + + wlf,fsampen: + description: FSAMPEN pin value, 0 for low, 1 for high, 2 for disconnected. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - Vdda-supply + - Vdd-supply + +unevaluatedProperties: false + +examples: + - | + wm8782: codec { + compatible = "wlf,wm8782"; + Vdda-supply = <&vdda_supply>; + Vdd-supply = <&vdd_supply>; + wlf,fsampen = <2>; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8804.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8804.yaml new file mode 100644 index 000000000..3c060179f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8804.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8804.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8804 audio codec + +description: | + This device supports both I2C and SPI (configured with pin strapping on the + board). + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8804 + + reg: + description: + The I2C address of the device for I2C, the chip select number for SPI. + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + PVDD-supply: + description: PLL core supply + + DVDD-supply: + description: Digital core supply + + wlf,reset-gpio: + description: A GPIO specifier for the GPIO controlling the reset pin. + maxItems: 1 + +required: + - reg + - compatible + - PVDD-supply + - DVDD-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8804"; + reg = <0x1a>; + PVDD-supply = <&pvdd_reg>; + DVDD-supply = <&dvdd_reg>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000..089b67384 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,115 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8903.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8903 audio codec + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + +maintainers: + - patches@opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8903 + + reg: + maxItems: 1 + + gpio-controller: true + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + micdet-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + description: Default register value for R6 (Mic Bias). + + micdet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 100 + description: The debounce delay for microphone detection in mS. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml new file mode 100644 index 000000000..3029a868e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml @@ -0,0 +1,203 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/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 + + wlf,in1l-as-dmicdat1: + type: boolean + description: + Use IN1L/DMICDAT1 as DMICDAT1, enabling the DMIC input path. + Can be used separately or together with wlf,in1r-as-dmicdat2. + + wlf,in1r-as-dmicdat2: + type: boolean + description: + Use IN1R/DMICDAT2 as DMICDAT2, enabling the DMIC input path. + Can be used separately or together with wlf,in1l-as-dmicdat1. + + wlf,gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 4 + maxItems: 4 + description: + Default register values for R121/122/123/124 (GPIO Control). + If any entry has the value 0xFFFF, the related register won't be set. + default: [0xFFFF, 0xFFFF, 0xFFFF, 0xFFFF] + + wlf,micbias-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + description: + Default register values for R6/R7 (Mic Bias Control). + default: [0, 0] + + wlf,drc-cfg-names: + $ref: /schemas/types.yaml#/definitions/string-array + description: + List of strings for the available DRC modes. + If absent, DRC is disabled. + + wlf,drc-cfg-regs: + $ref: /schemas/types.yaml#/definitions/uint16-matrix + description: + Sets of default register values for R40/41/42/43 (DRC). + Each set corresponds to a DRC mode, so the number of sets should equal + the length of wlf,drc-cfg-names. + If absent, DRC is disabled. + items: + minItems: 4 + maxItems: 4 + + wlf,retune-mobile-cfg-names: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + List of strings for the available retune modes. + If absent, retune is disabled. + + wlf,retune-mobile-cfg-hz: + description: + The list must be the same length as wlf,retune-mobile-cfg-names. + If absent, retune is disabled. + + wlf,retune-mobile-cfg-regs: + $ref: /schemas/types.yaml#/definitions/uint16-matrix + description: + Sets of default register values for R134/.../157 (EQ). + Each set corresponds to a retune mode, so the number of sets should equal + the length of wlf,retune-mobile-cfg-names. + If absent, retune is disabled. + items: + minItems: 24 + maxItems: 24 + +dependencies: + wlf,drc-cfg-names: [ 'wlf,drc-cfg-regs' ] + wlf,drc-cfg-regs: [ 'wlf,drc-cfg-names' ] + + wlf,retune-mobile-cfg-names: [ 'wlf,retune-mobile-cfg-hz', 'wlf,retune-mobile-cfg-regs' ] + wlf,retune-mobile-cfg-regs: [ 'wlf,retune-mobile-cfg-names', 'wlf,retune-mobile-cfg-hz' ] + wlf,retune-mobile-cfg-hz: [ 'wlf,retune-mobile-cfg-names', 'wlf,retune-mobile-cfg-regs' ] + +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>; + + wlf,drc-cfg-names = "default", "peaklimiter", "tradition", "soft", + "music"; + /* + * Config registers per name, respectively: + * KNEE_IP = 0, KNEE_OP = 0, HI_COMP = 1, LO_COMP = 1 + * KNEE_IP = -24, KNEE_OP = -6, HI_COMP = 1/4, LO_COMP = 1 + * KNEE_IP = -42, KNEE_OP = -3, HI_COMP = 0, LO_COMP = 1 + * KNEE_IP = -45, KNEE_OP = -9, HI_COMP = 1/8, LO_COMP = 1 + * KNEE_IP = -30, KNEE_OP = -10.5, HI_COMP = 1/4, LO_COMP = 1 + */ + wlf,drc-cfg-regs = /bits/ 16 <0x01af 0x3248 0x0000 0x0000>, + /bits/ 16 <0x04af 0x324b 0x0010 0x0408>, + /bits/ 16 <0x04af 0x324b 0x0028 0x0704>, + /bits/ 16 <0x04af 0x324b 0x0018 0x078c>, + /bits/ 16 <0x04af 0x324b 0x0010 0x050e>; + + /* GPIO1 = DMIC_CLK, don't touch others */ + wlf,gpio-cfg = <0x0018>, <0xffff>, <0xffff>, <0xffff>; + + /* Use IN1R as DMICDAT2, leave IN1L as an analog input path */ + wlf,in1r-as-dmicdat2; + + wlf,retune-mobile-cfg-names = "bassboost", "bassboost", "treble"; + wlf,retune-mobile-cfg-hz = <48000>, <44100>, <48000>; + /* + * Config registers per name, respectively: + * EQ_ENA, 100 Hz, 300 Hz, 875 Hz, 2400 Hz, 6900 Hz + * 1, +6 dB, +3 dB, 0 dB, 0 dB, 0 dB + * 1, +6 dB, +3 dB, 0 dB, 0 dB, 0 dB + * 1, -2 dB, -2 dB, 0 dB, 0 dB, +3 dB + * Each one uses the defaults for ReTune Mobile registers 140-157 + */ + wlf,retune-mobile-cfg-regs = /bits/ 16 <0x1 0x12 0xf 0xc 0xc 0xc + 0x0fca 0x0400 0x00d8 0x1eb5 + 0xf145 0x0bd5 0x0075 0x1c58 + 0xf3d3 0x0a54 0x0568 0x168e + 0xf829 0x07ad 0x1103 0x0564 + 0x0559 0x4000>, + + /bits/ 16 <0x1 0x12 0xf 0xc 0xc 0xc + 0x0fca 0x0400 0x00d8 0x1eb5 + 0xf145 0x0bd5 0x0075 0x1c58 + 0xf3d3 0x0a54 0x0568 0x168e + 0xf829 0x07ad 0x1103 0x0564 + 0x0559 0x4000>, + + /bits/ 16 <0x1 0xa 0xa 0xc 0xc 0xf + 0x0fca 0x0400 0x00d8 0x1eb5 + 0xf145 0x0bd5 0x0075 0x1c58 + 0xf3d3 0x0a54 0x0568 0x168e + 0xf829 0x07ad 0x1103 0x0564 + 0x0559 0x4000>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml new file mode 100644 index 000000000..3e809217c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8940.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8940 Codec + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8940 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml new file mode 100644 index 000000000..c8c786cb6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml @@ -0,0 +1,134 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$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 + +description: | + Wolfson WM8960 audio codec + + Pins on the device (for linking into audio routes): + + Outputs: + * HP_L : Left Headphone/Line Output + * HP_R : Right Headphone/Line Output + * SPK_LP : Left Speaker Output (Positive) + * SPK_LN : Left Speaker Output (Negative) + * SPK_RP : Right Speaker Output (Positive) + * SPK_RN : Right Speaker Output (Negative) + * OUT3 : Mono, Left, Right or buffered midrail output for capless mode + + Inputs: + * LINPUT1 : Left single-ended or negative differential microphone input + * RINPUT1 : Right single-ended or negative differential microphone input + * LINPUT2 : Left line input or positive differential microphone input + * RINPUT2 : Right line input or positive differential microphone input + * LINPUT3 : Left line input, positive differential microphone, or Jack Detect 2 + * RINPUT3 : Right line input, positive differential microphone, or Jack Detect 3 + +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. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + clocks = <&clks 0>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + wlf,hp-cfg = <3 2 3>; + wlf,gpio-cfg = <1 3>; + wlf,shared-lrclk; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml new file mode 100644 index 000000000..5fe0b2c9f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml @@ -0,0 +1,124 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8962.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8962 Ultra-Low Power Stereo CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8962 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + interrupts: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + AVDD-supply: + description: Analogue supply. + + CPVDD-supply: + description: Charge pump power supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + MICVDD-supply: + description: Microphone bias amp supply. + + PLLVDD-supply: + description: PLL Supply + + SPKVDD1-supply: + description: Supply for left speaker drivers. + + SPKVDD2-supply: + description: Supply for right speaker drivers. + + spk-mono: + $ref: /schemas/types.yaml#/definitions/flag + description: + If present, the SPK_MONO bit of R51 (Class D Control 2) gets set, + indicating that the speaker is in mono mode. + + mic-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Default register value for R48 (Additional Control 4). + If absent, the default should be the register default. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + description: + A list of GPIO configuration register values. If absent, no + configuration of these registers is performed. Note that only values + within [0x0, 0xffff] are valid. Any other value is regarded as setting + the GPIO register to its reset value 0x0. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + - PLLVDD-supply + - SPKVDD1-supply + - SPKVDD2-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/imx6qdl-clock.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8962: codec@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; + clocks = <&clks IMX6QDL_CLK_CKO>; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + CPVDD-supply = <®_audio>; + MICVDD-supply = <®_audio>; + PLLVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; + gpio-cfg = < + 0x0000 /* 0:Default */ + 0x0000 /* 1:Default */ + 0x0013 /* 2:FN_DMICCLK */ + 0x0000 /* 3:Default */ + 0x8014 /* 4:FN_DMICCDAT */ + 0x0000 /* 5:Default */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000..efb5f9f6c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0x1a>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8994.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8994.yaml new file mode 100644 index 000000000..0db04a90a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8994.yaml @@ -0,0 +1,194 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8994.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM1811/WM8994/WM8958 audio codecs + +maintainers: + - Krzysztof Kozlowski <krzk@kernel.org> + - patches@opensource.cirrus.com + +description: | + These devices support both I2C and SPI (configured with pin strapping on the + board). + + Pins on the device (for linking into audio routes): + IN1LN, IN1LP, IN2LN, IN2LP:VXRN, IN1RN, IN1RP, IN2RN, IN2RP:VXRP, SPKOUTLP, + SPKOUTLN, SPKOUTRP, SPKOUTRN, HPOUT1L, HPOUT1R, HPOUT2P, HPOUT2N, LINEOUT1P, + LINEOUT1N, LINEOUT2P, LINEOUT2N. + +properties: + compatible: + enum: + - wlf,wm1811 + - wlf,wm8994 + - wlf,wm8958 + + reg: + maxItems: 1 + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + minItems: 1 + items: + - const: MCLK1 + - const: MCLK2 + + gpio-controller: true + + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + interrupt-controller: true + + '#interrupt-cells': + const: 2 + description: + The first cell is the IRQ number. The second cell is the flags, encoded + as the trigger masks. + + AVDD1-supply: true + AVDD2-supply: true + CPVDD-supply: true + DBVDD-supply: true + DBVDD1-supply: true + DBVDD2-supply: true + DBVDD3-supply: true + DCVDD-supply: true + LDO1VDD-supply: true + LDO2VDD-supply: true + SPKVDD1-supply: true + SPKVDD2-supply: true + + '#sound-dai-cells': + const: 0 + + wlf,gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 11 + description: + A list of GPIO configuration register values. If absent, no configuration + of these registers is performed. If any value is over 0xffff then the + register will be left as default. If present 11 values must be supplied. + + wlf,micbias-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + description: + Two MICBIAS register values for WM1811 or WM8958. If absent the register + defaults will be used. + + wlf,ldo1ena-gpios: + maxItems: 1 + description: + Control of LDO1ENA input to device. + + wlf,ldo2ena-gpios: + maxItems: 1 + description: + Control of LDO2ENA input to device. + + wlf,lineout1-se: + type: boolean + description: + LINEOUT1 is in single ended mode. + + wlf,lineout2-se: + type: boolean + description: + INEOUT2 is in single ended mode. + + wlf,lineout1-feedback: + type: boolean + description: + LINEOUT1 has common mode feedback connected. + + wlf,lineout2-feedback: + type: boolean + description: + LINEOUT2 has common mode feedback connected. + + wlf,ldoena-always-driven: + type: boolean + description: + LDOENA is always driven. + + wlf,spkmode-pu: + type: boolean + description: + Enable the internal pull-up resistor on the SPKMODE pin. + + wlf,csnaddr-pd: + type: boolean + description: + Enable the internal pull-down resistor on the CS/ADDR pin. + +required: + - compatible + - reg + - AVDD2-supply + - CPVDD-supply + - SPKVDD1-supply + - SPKVDD2-supply + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + enum: + - wlf,wm1811 + - wlf,wm8958 + then: + properties: + DBVDD-supply: false + LDO2VDD-supply: false + required: + - DBVDD1-supply + - DBVDD2-supply + - DBVDD3-supply + else: + properties: + DBVDD1-supply: false + DBVDD2-supply: false + DBVDD3-supply: false + required: + - DBVDD-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1a { + compatible = "wlf,wm1811"; + reg = <0x1a>; + clocks = <&i2s0 0>; + clock-names = "MCLK1"; + + AVDD2-supply = <&main_dc_reg>; + CPVDD-supply = <&main_dc_reg>; + DBVDD1-supply = <&main_dc_reg>; + DBVDD2-supply = <&main_dc_reg>; + DBVDD3-supply = <&main_dc_reg>; + LDO1VDD-supply = <&main_dc_reg>; + SPKVDD1-supply = <&main_dc_reg>; + SPKVDD2-supply = <&main_dc_reg>; + + wlf,ldo1ena-gpios = <&gpb0 0 GPIO_ACTIVE_HIGH>; + wlf,ldo2ena-gpios = <&gpb0 1 GPIO_ACTIVE_HIGH>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt new file mode 100644 index 000000000..b69e196c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -0,0 +1,29 @@ +WM8741 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8741" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + +Example: + +wm8741: codec@1a { + compatible = "wlf,wm8741"; + reg = <0x1a>; + + diff-mode = <3>; +}; diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.yaml b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.yaml new file mode 100644 index 000000000..82fa448bd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/xlnx,audio-formatter.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Xilinx PL audio formatter + +description: + The IP core supports DMA, data formatting(AES<->PCM conversion) + of audio samples. + +maintainers: + - Vincenzo Frascino <vincenzo.frascino@arm.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - xlnx,audio-formatter-1.0 + + reg: + maxItems: 1 + + interrupt-names: + minItems: 1 + items: + - const: irq_mm2s + - const: irq_s2mm + + interrupts: + minItems: 1 + items: + - description: interrupt from MM2S block + - description: interrupt from S2MM block + + clock-names: + minItems: 1 + items: + - const: s_axi_lite_aclk + - const: aud_mclk + + clocks: + minItems: 1 + items: + - description: clock for the axi data stream + - description: clock for the MEMS microphone data stream + +required: + - compatible + - reg + - interrupt-names + - interrupts + - clock-names + - clocks + +additionalProperties: false + +examples: + - | + audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + reg = <0x80010000 0x1000>; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/xlnx,i2s.yaml b/Documentation/devicetree/bindings/sound/xlnx,i2s.yaml new file mode 100644 index 000000000..3c2b0be07 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,i2s.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/xlnx,i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Xilinx I2S PL block + +description: + The IP supports I2S based playback/capture audio. + +maintainers: + - Vincenzo Frascino <vincenzo.frascino@arm.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - xlnx,i2s-receiver-1.0 + - xlnx,i2s-transmitter-1.0 + + reg: + maxItems: 1 + + xlnx,dwidth: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 16 + - 24 + description: | + Sample data width. + + xlnx,num-channels: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 4 + description: | + Number of I2S streams. + +required: + - compatible + - reg + - xlnx,dwidth + - xlnx,num-channels + +additionalProperties: false + +examples: + - | + i2s@a0080000 { + compatible = "xlnx,i2s-receiver-1.0"; + reg = <0xa0080000 0x10000>; + xlnx,dwidth = <0x18>; + xlnx,num-channels = <1>; + }; + i2s@a0090000 { + compatible = "xlnx,i2s-transmitter-1.0"; + reg = <0xa0090000 0x10000>; + xlnx,dwidth = <0x18>; + xlnx,num-channels = <1>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/xlnx,spdif.yaml b/Documentation/devicetree/bindings/sound/xlnx,spdif.yaml new file mode 100644 index 000000000..a45d8a075 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,spdif.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/xlnx,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Xilinx SPDIF IP + +description: + The IP supports playback and capture of SPDIF audio. + +maintainers: + - Vincenzo Frascino <vincenzo.frascino@arm.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - xlnx,spdif-2.0 + + reg: + maxItems: 1 + + interrupts: + items: + - description: SPDIF audio interrupt + + clock-names: + items: + - const: aud_clk_i + - const: s_axi_aclk + + clocks: + minItems: 1 + items: + - description: input audio clock + - description: clock for the AXI data stream + + xlnx,spdif-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 + - 1 + description: | + 0 - receiver + 1 - transmitter + + xlnx,aud_clk_i: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Input audio clock frequency. It affects the sampling rate. + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + +additionalProperties: false + +examples: + - | + spdif@80010000 { + compatible = "xlnx,spdif-2.0"; + reg = <0x80010000 0x10000>; + clock-names = "aud_clk_i", "s_axi_aclk"; + clocks = <&misc_clk_0>, <&clk 71>; + interrupt-parent = <&gic>; + interrupts = <0 91 4>; + xlnx,spdif-mode = <1>; + xlnx,aud_clk_i = <49152913>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/xmos,xvf3500.yaml b/Documentation/devicetree/bindings/sound/xmos,xvf3500.yaml new file mode 100644 index 000000000..fb77a61f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xmos,xvf3500.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/xmos,xvf3500.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: XMOS XVF3500 VocalFusion Voice Processor + +maintainers: + - Javier Carrasco <javier.carrasco@wolfvision.net> + +description: + The XMOS XVF3500 VocalFusion Voice Processor is a low-latency, 32-bit + multicore controller for voice processing. + https://www.xmos.com/xvf3500/ + +allOf: + - $ref: /schemas/usb/usb-device.yaml# + +properties: + compatible: + const: usb20b1,0013 + + reg: true + + reset-gpios: + maxItems: 1 + + vdd-supply: + description: + Regulator for the 1V0 supply. + + vddio-supply: + description: + Regulator for the 3V3 supply. + +required: + - compatible + - reg + - reset-gpios + - vdd-supply + - vddio-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + usb { + #address-cells = <1>; + #size-cells = <0>; + + voice_processor: voice-processor@1 { + compatible = "usb20b1,0013"; + reg = <1>; + reset-gpios = <&gpio 5 GPIO_ACTIVE_LOW>; + vdd-supply = <&vcc1v0>; + vddio-supply = <&vcc3v3>; + }; + }; + +... |