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authorTakashi Iwai <tiwai@suse.de>2024-09-14 09:09:59 +0200
committerTakashi Iwai <tiwai@suse.de>2024-09-14 09:09:59 +0200
commit1a529af6f81e54f15df162a0c703459937941c54 (patch)
treeb4ff93cec914dfa923fe2c3b05bc1faf71c83e10 /Documentation/devicetree/bindings
parent5740434e1e0f51db1282436a7783658e6c139fd1 (diff)
parent2772ee6de6cf94e5f2a0c0ce6067d0796a4170ba (diff)
Merge tag 'asoc-v6.12' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.12 This is a very large set of changes, almost all in drivers rather than the core. Even with the addition of several quite large drivers the overall diffstat is negative thanks to the removal of some old Intel board support which has been obsoleted by the AVS driver, helped a bit by some factoring out into helpers (especially around the Soundwire machine drivers for x86). Highlights include: - More simplifications and cleanups throughout the subsystem from Morimoto-san. - Extensive cleanups and refactoring of the Soundwire drivers to make better use of helpers. - Removal of Intel machine support obsoleted by the AVS driver. - Lots of DT schema conversions. - Machine support for many AMD and Intel x86 platforms. - Support for AMD ACP 7.1, Mediatek MT6367 and MT8365, Realtek RTL1320 SoundWire and rev C, and Texas Instruments TAS2563
Diffstat (limited to 'Documentation/devicetree/bindings')
-rw-r--r--Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml21
-rw-r--r--Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml7
-rw-r--r--Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml7
-rw-r--r--Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml101
-rw-r--r--Documentation/devicetree/bindings/sound/cs4271.txt57
-rw-r--r--Documentation/devicetree/bindings/sound/da7213.txt45
-rw-r--r--Documentation/devicetree/bindings/sound/dlg,da7213.yaml103
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml111
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,saif.yaml83
-rw-r--r--Documentation/devicetree/bindings/sound/imx-audio-es8328.txt60
-rw-r--r--Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml130
-rw-r--r--Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml107
-rw-r--r--Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml5
-rw-r--r--Documentation/devicetree/bindings/sound/mxs-saif.txt41
-rw-r--r--Documentation/devicetree/bindings/sound/pcm512x.txt53
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml205
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml22
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,sm8250.yaml137
-rw-r--r--Documentation/devicetree/bindings/sound/realtek,rt5616.yaml12
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.yaml6
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml4
-rw-r--r--Documentation/devicetree/bindings/sound/samsung,odroid.yaml5
-rw-r--r--Documentation/devicetree/bindings/sound/ti,pcm512x.yaml101
-rw-r--r--Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml127
-rw-r--r--Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml55
-rw-r--r--Documentation/devicetree/bindings/sound/tlv320aic31xx.txt77
-rw-r--r--Documentation/devicetree/bindings/sound/tpa6130a2.txt27
27 files changed, 1186 insertions, 523 deletions
diff --git a/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml b/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml
index 37423c2e0fdf..b67fbe0e7a63 100644
--- a/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml
+++ b/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml
@@ -37,6 +37,24 @@ properties:
"#interrupt-cells":
const: 2
+ mediatek,hp-pull-down:
+ description:
+ Earphone driver positive output stage short to
+ the audio reference ground.
+ type: boolean
+
+ mediatek,micbias0-microvolt:
+ description: Selects MIC Bias 0 output voltage.
+ enum: [1700000, 1800000, 1900000, 2000000,
+ 2100000, 2500000, 2600000, 2700000]
+ default: 1700000
+
+ mediatek,micbias1-microvolt:
+ description: Selects MIC Bias 1 output voltage.
+ enum: [1700000, 1800000, 1900000, 2000000,
+ 2100000, 2500000, 2600000, 2700000]
+ default: 1700000
+
regulators:
type: object
$ref: /schemas/regulator/mediatek,mt6357-regulator.yaml
@@ -83,6 +101,9 @@ examples:
interrupt-controller;
#interrupt-cells = <2>;
+ mediatek,micbias0-microvolt = <1700000>;
+ mediatek,micbias1-microvolt = <1700000>;
+
regulators {
mt6357_vproc_reg: buck-vproc {
regulator-name = "vproc";
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml
index 5db718e4d0e7..4f13e8ab50b2 100644
--- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml
@@ -26,6 +26,13 @@ properties:
A list off component DAPM widget. Each entry is a pair of strings,
the first being the widget type, the second being the widget name
+ clocks:
+ minItems: 1
+ maxItems: 3
+ description:
+ Base PLL clocks of audio susbsytem, used to configure base clock
+ frequencies for different audio use-cases.
+
patternProperties:
"^dai-link-[0-9]+$":
type: object
diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml
index 0ecdaf7190e9..413b47778181 100644
--- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml
+++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml
@@ -27,6 +27,13 @@ properties:
A list off component DAPM widget. Each entry is a pair of strings,
the first being the widget type, the second being the widget name
+ clocks:
+ minItems: 1
+ maxItems: 3
+ description:
+ Base PLL clocks of audio susbsytem, used to configure base clock
+ frequencies for different audio use-cases.
+
patternProperties:
"^dai-link-[0-9]+$":
type: object
diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml
new file mode 100644
index 000000000000..68fbf5cc208f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml
@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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
+
+ 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/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt
deleted file mode 100644
index 6e699ceabacd..000000000000
--- a/Documentation/devicetree/bindings/sound/cs4271.txt
+++ /dev/null
@@ -1,57 +0,0 @@
-Cirrus Logic CS4271 DT bindings
-
-This driver supports both the I2C and the SPI bus.
-
-Required properties:
-
- - compatible: "cirrus,cs4271"
-
-For required properties on SPI, please consult
-Documentation/devicetree/bindings/spi/spi-bus.txt
-
-Required properties on I2C:
-
- - reg: the i2c address
-
-
-Optional properties:
-
- - reset-gpio: a GPIO spec to define which pin is connected to the chip's
- !RESET pin
- - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag
- is enabled.
- - cirrus,enable-soft-reset:
- The CS4271 requires its LRCLK and MCLK to be stable before its RESET
- line is de-asserted. That also means that clocks cannot be changed
- without putting the chip back into hardware reset, which also requires
- a complete re-initialization of all registers.
-
- One (undocumented) workaround is to assert and de-assert the PDN bit
- in the MODE2 register. This workaround can be enabled with this DT
- property.
-
- Note that this is not needed in case the clocks are stable
- throughout the entire runtime of the codec.
-
- - vd-supply: Digital power
- - vl-supply: Logic power
- - va-supply: Analog Power
-
-Examples:
-
- codec_i2c: cs4271@10 {
- compatible = "cirrus,cs4271";
- reg = <0x10>;
- reset-gpio = <&gpio 23 0>;
- vd-supply = <&vdd_3v3_reg>;
- vl-supply = <&vdd_3v3_reg>;
- va-supply = <&vdd_3v3_reg>;
- };
-
- codec_spi: cs4271@0 {
- compatible = "cirrus,cs4271";
- reg = <0x0>;
- reset-gpio = <&gpio 23 0>;
- spi-max-frequency = <6000000>;
- };
-
diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt
deleted file mode 100644
index 94584c96c4ae..000000000000
--- a/Documentation/devicetree/bindings/sound/da7213.txt
+++ /dev/null
@@ -1,45 +0,0 @@
-Dialog Semiconductor DA7212/DA7213 Audio Codec bindings
-
-======
-
-Required properties:
-- compatible : Should be "dlg,da7212" or "dlg,da7213"
-- reg: Specifies the I2C slave address
-
-Optional properties:
-- clocks : phandle and clock specifier for codec MCLK.
-- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
-
-- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1
- [<1600>, <2200>, <2500>, <3000>]
-- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2
- [<1600>, <2200>, <2500>, <3000>]
-- dlg,dmic-data-sel : DMIC channel select based on clock edge.
- ["lrise_rfall", "lfall_rrise"]
-- dlg,dmic-samplephase : When to sample audio from DMIC.
- ["on_clkedge", "between_clkedge"]
-- dlg,dmic-clkrate : DMIC clock frequency (Hz).
- [<1500000>, <3000000>]
-
- - VDDA-supply : Regulator phandle for Analogue power supply
- - VDDMIC-supply : Regulator phandle for Mic Bias
- - VDDIO-supply : Regulator phandle for I/O power supply
-
-======
-
-Example:
-
- codec_i2c: da7213@1a {
- compatible = "dlg,da7213";
- reg = <0x1a>;
-
- clocks = <&clks 201>;
- clock-names = "mclk";
-
- dlg,micbias1-lvl = <2500>;
- dlg,micbias2-lvl = <2500>;
-
- dlg,dmic-data-sel = "lrise_rfall";
- dlg,dmic-samplephase = "between_clkedge";
- dlg,dmic-clkrate = <3000000>;
- };
diff --git a/Documentation/devicetree/bindings/sound/dlg,da7213.yaml b/Documentation/devicetree/bindings/sound/dlg,da7213.yaml
new file mode 100644
index 000000000000..c2dede1e82ff
--- /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:
+ $ref: audio-graph-port.yaml#
+ unevaluatedProperties: false
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "dlg,da7213";
+ reg = <0x1a>;
+
+ clocks = <&clks 201>;
+ clock-names = "mclk";
+
+ #sound-dai-cells = <0>;
+
+ dlg,micbias1-lvl = <2500>;
+ dlg,micbias2-lvl = <2500>;
+
+ dlg,dmic-data-sel = "lrise_rfall";
+ dlg,dmic-samplephase = "between_clkedge";
+ dlg,dmic-clkrate = <3000000>;
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml
new file mode 100644
index 000000000000..5eb6f5812cf2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml
@@ -0,0 +1,111 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: The external port of the i.MX audio muxer (AUDMIX)
+ enum: [3, 4, 5, 6]
+ default: 3
+
+required:
+ - compatible
+ - model
+ - ssi-controller
+ - jack-gpio
+ - audio-amp-supply
+ - audio-codec
+ - audio-routing
+ - mux-int-port
+ - mux-ext-port
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "fsl,imx-audio-es8328";
+ model = "imx-audio-es8328";
+ ssi-controller = <&ssi1>;
+ audio-codec = <&codec>;
+ jack-gpio = <&gpio5 15 0>;
+ audio-amp-supply = <&reg_audio_amp>;
+ audio-routing =
+ "Speaker", "LOUT2",
+ "Speaker", "ROUT2",
+ "Speaker", "audio-amp",
+ "Headphone", "ROUT1",
+ "Headphone", "LOUT1",
+ "LINPUT1", "Mic Jack",
+ "RINPUT1", "Mic Jack",
+ "Mic Jack", "Mic Bias";
+ mux-int-port = <1>;
+ mux-ext-port = <3>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/fsl,saif.yaml b/Documentation/devicetree/bindings/sound/fsl,saif.yaml
new file mode 100644
index 000000000000..0b5db6bb1b7c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,saif.yaml
@@ -0,0 +1,83 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
deleted file mode 100644
index 07b68ab206fb..000000000000
--- a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
+++ /dev/null
@@ -1,60 +0,0 @@
-Freescale i.MX audio complex with ES8328 codec
-
-Required properties:
-- compatible : "fsl,imx-audio-es8328"
-- model : The user-visible name of this sound complex
-- ssi-controller : The phandle of the i.MX SSI controller
-- jack-gpio : Optional GPIO for headphone jack
-- audio-amp-supply : Power regulator for speaker amps
-- audio-codec : The phandle of the ES8328 audio codec
-- audio-routing : A list of the connections between audio components.
- Each entry is a pair of strings, the first being the
- connection's sink, the second being the connection's
- source. Valid names could be power supplies, ES8328
- pins, and the jacks on the board:
-
- Power supplies:
- * audio-amp
-
- ES8328 pins:
- * LOUT1
- * LOUT2
- * ROUT1
- * ROUT2
- * LINPUT1
- * LINPUT2
- * RINPUT1
- * RINPUT2
- * Mic PGA
-
- Board connectors:
- * Headphone
- * Speaker
- * Mic Jack
-- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
-- mux-ext-port : The external port of the i.MX audio muxer (AUDMIX)
-
-Note: The AUDMUX port numbering should start at 1, which is consistent with
-hardware manual.
-
-Example:
-
-sound {
- compatible = "fsl,imx-audio-es8328";
- model = "imx-audio-es8328";
- ssi-controller = <&ssi1>;
- audio-codec = <&codec>;
- jack-gpio = <&gpio5 15 0>;
- audio-amp-supply = <&reg_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/mediatek,mt8365-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml
new file mode 100644
index 000000000000..45ad56d37234
--- /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 000000000000..ff9ebb63a05f
--- /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/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml
index 2f43c684ab88..7fbab5871be4 100644
--- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml
+++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml
@@ -13,6 +13,9 @@ 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
@@ -53,7 +56,7 @@ required:
- dmas
- dma-names
-additionalProperties: false
+unevaluatedProperties: false
examples:
- |
diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt
deleted file mode 100644
index 7ba07a118e37..000000000000
--- a/Documentation/devicetree/bindings/sound/mxs-saif.txt
+++ /dev/null
@@ -1,41 +0,0 @@
-* Freescale MXS Serial Audio Interface (SAIF)
-
-Required properties:
-- compatible: Should be "fsl,<chip>-saif"
-- reg: Should contain registers location and length
-- interrupts: Should contain ERROR interrupt number
-- dmas: DMA specifier, consisting of a phandle to DMA controller node
- and SAIF DMA channel ID.
- Refer to dma.txt and fsl-mxs-dma.txt for details.
-- dma-names: Must be "rx-tx".
-
-Optional properties:
-- fsl,saif-master: phandle to the master SAIF. It's only required for
- the slave SAIF.
-
-Note: Each SAIF controller should have an alias correctly numbered
-in "aliases" node.
-
-Example:
-
-aliases {
- saif0 = &saif0;
- saif1 = &saif1;
-};
-
-saif0: saif@80042000 {
- compatible = "fsl,imx28-saif";
- reg = <0x80042000 2000>;
- interrupts = <59>;
- dmas = <&dma_apbx 4>;
- dma-names = "rx-tx";
-};
-
-saif1: saif@80046000 {
- compatible = "fsl,imx28-saif";
- reg = <0x80046000 2000>;
- interrupts = <58>;
- dmas = <&dma_apbx 5>;
- dma-names = "rx-tx";
- fsl,saif-master = <&saif0>;
-};
diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt
deleted file mode 100644
index 47878a6df608..000000000000
--- a/Documentation/devicetree/bindings/sound/pcm512x.txt
+++ /dev/null
@@ -1,53 +0,0 @@
-PCM512x and TAS575x audio CODECs/amplifiers
-
-These devices support both I2C and SPI (configured with pin strapping
-on the board). The TAS575x devices only support I2C.
-
-Required properties:
-
- - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141",
- "ti,pcm5142", "ti,pcm5242", "ti,tas5754" or "ti,tas5756"
-
- - reg : the I2C address of the device for I2C, the chip select
- number for SPI.
-
- - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the
- device, as covered in bindings/regulator/regulator.txt
-
-Optional properties:
-
- - clocks : A clock specifier for the clock connected as SCLK. If this
- is absent the device will be configured to clock from BCLK. If pll-in
- and pll-out are specified in addition to a clock, the device is
- configured to accept clock input on a specified gpio pin.
-
- - pll-in, pll-out : gpio pins used to connect the pll using <1>
- through <6>. The device will be configured for clock input on the
- given pll-in pin and PLL output on the given pll-out pin. An
- external connection from the pll-out pin to the SCLK pin is assumed.
- Caution: the TAS-desvices only support gpios 1,2 and 3
-
-Examples:
-
- pcm5122: pcm5122@4c {
- compatible = "ti,pcm5122";
- reg = <0x4c>;
-
- AVDD-supply = <&reg_3v3_analog>;
- DVDD-supply = <&reg_1v8>;
- CPVDD-supply = <&reg_3v3>;
- };
-
-
- pcm5142: pcm5142@4c {
- compatible = "ti,pcm5142";
- reg = <0x4c>;
-
- AVDD-supply = <&reg_3v3_analog>;
- DVDD-supply = <&reg_1v8>;
- CPVDD-supply = <&reg_3v3>;
-
- clocks = <&sck>;
- pll-in = <3>;
- pll-out = <6>;
- };
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 000000000000..6ad451549036
--- /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-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml
index 06b5f7be3608..6f5644a89feb 100644
--- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml
@@ -64,6 +64,7 @@ allOf:
compatible:
enum:
- qcom,sc7280-lpass-wsa-macro
+ - qcom,sm8250-lpass-wsa-macro
- qcom,sm8450-lpass-wsa-macro
- qcom,sc8280xp-lpass-wsa-macro
then:
@@ -82,24 +83,6 @@ allOf:
- if:
properties:
compatible:
- enum:
- - qcom,sm8250-lpass-wsa-macro
- then:
- properties:
- clocks:
- minItems: 6
- clock-names:
- items:
- - const: mclk
- - const: npl
- - const: macro
- - const: dcodec
- - const: va
- - const: fsgen
-
- - if:
- properties:
- compatible:
contains:
enum:
- qcom,sm8550-lpass-wsa-macro
@@ -130,8 +113,7 @@ examples:
<&audiocc 0>,
<&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
<&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
- <&aoncc LPASS_CDC_VA_MCLK>,
<&vamacro>;
- clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen";
+ clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
clock-output-names = "mclk";
};
diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
index c9076dcd44c1..1d3acdc0c733 100644
--- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
@@ -27,9 +27,7 @@ properties:
- qcom,sm8650-sndcard
- const: qcom,sm8450-sndcard
- enum:
- - qcom,apq8016-sbc-sndcard
- qcom,apq8096-sndcard
- - qcom,msm8916-qdsp6-sndcard
- qcom,qcm6490-idp-sndcard
- qcom,qcs6490-rb3gen2-sndcard
- qcom,qrb5165-rb5-sndcard
@@ -58,18 +56,6 @@ properties:
$ref: /schemas/types.yaml#/definitions/string
description: User visible long sound card name
- pin-switches:
- description: List of widget names for which pin switches should be created.
- $ref: /schemas/types.yaml#/definitions/string-array
-
- widgets:
- description: User specified audio sound widgets.
- $ref: /schemas/types.yaml#/definitions/non-unique-string-array
-
- # Only valid for some compatibles (see allOf if below)
- reg: true
- reg-names: true
-
patternProperties:
".*-dai-link$":
description:
@@ -122,34 +108,6 @@ required:
- compatible
- model
-allOf:
- - if:
- properties:
- compatible:
- contains:
- enum:
- - qcom,apq8016-sbc-sndcard
- - qcom,msm8916-qdsp6-sndcard
- then:
- properties:
- reg:
- items:
- - description: Microphone I/O mux register address
- - description: Speaker I/O mux register address
- reg-names:
- items:
- - const: mic-iomux
- - const: spkr-iomux
- required:
- - compatible
- - model
- - reg
- - reg-names
- else:
- properties:
- reg: false
- reg-names: false
-
additionalProperties: false
examples:
@@ -231,98 +189,3 @@ 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/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml
index 248320804e5f..29071044c66e 100644
--- a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml
+++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml
@@ -30,6 +30,18 @@ properties:
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
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index 07ec6247d9de..3bc93c59535e 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -112,6 +112,12 @@ properties:
description: List of necessary clock names.
# details are defined below
+ post-init-providers:
+ description: At least if rsnd is using DPCM connection on Audio-Graph-Card2,
+ fw_devlink might doesn't have enough information to break the cycle. rsnd
+ driver will not be probed in such case. Same problem might occur with
+ Multi-CPU/Codec or Codec2Codec.
+
# ports is below
port:
$ref: audio-graph-port.yaml#/definitions/port-base
diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml
index 8b9695f5decc..f4610eaed1e1 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml
@@ -87,6 +87,10 @@ properties:
'#sound-dai-cells':
const: 0
+ port:
+ $ref: audio-graph-port.yaml#/definitions/port-base
+ description: Connection to controller providing I2S signals
+
required:
- compatible
- reg
diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml
index b77284e3e26a..c3dea852cc8d 100644
--- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml
+++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml
@@ -27,11 +27,6 @@ properties:
- const: samsung,odroid-xu4-audio
deprecated: true
- assigned-clock-parents: true
- assigned-clock-rates: true
- assigned-clocks: true
- clocks: true
-
cpu:
type: object
additionalProperties: false
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml b/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml
new file mode 100644
index 000000000000..21ea9ff5a2bb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml
@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ minimum: 1
+ maximum: 6
+
+required:
+ - compatible
+ - reg
+ - AVDD-supply
+ - DVDD-supply
+ - CPVDD-supply
+
+if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - ti,tas5754
+ - ti,tas5756
+
+then:
+ properties:
+ pll-in:
+ maximum: 3
+
+ pll-out:
+ maximum: 3
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@4c {
+ compatible = "ti,pcm5142";
+ reg = <0x4c>;
+ AVDD-supply = <&reg_3v3_analog>;
+ DVDD-supply = <&reg_1v8>;
+ CPVDD-supply = <&reg_3v3>;
+ #sound-dai-cells = <0>;
+ clocks = <&sck>;
+ pll-in = <3>;
+ pll-out = <6>;
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml
new file mode 100644
index 000000000000..85e937e34962
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml
@@ -0,0 +1,127 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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:
+ - $ref: dai-common.yaml#
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ #include <dt-bindings/sound/tlv320aic31xx.h>
+
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ sound@18 {
+ compatible = "ti,tlv320aic311x";
+ reg = <0x18>;
+
+ ai31xx-micbias-vg = <MICBIAS_2_0V>;
+ reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
+ HPVDD-supply = <&regulator>;
+ SPRVDD-supply = <&regulator>;
+ SPLVDD-supply = <&regulator>;
+ AVDD-supply = <&regulator>;
+ IOVDD-supply = <&regulator>;
+ DVDD-supply = <&regulator>;
+ };
+ };
+
diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml
new file mode 100644
index 000000000000..a42bf9bde694
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml
@@ -0,0 +1,55 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$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/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
deleted file mode 100644
index bbad98d5b986..000000000000
--- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
+++ /dev/null
@@ -1,77 +0,0 @@
-Texas Instruments - tlv320aic31xx Codec module
-
-The tlv320aic31xx serial control bus communicates through I2C protocols
-
-Required properties:
-
-- compatible - "string" - One of:
- "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp
- "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp
- "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP)
- "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
- "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
- "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
- "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP)
- "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP)
-
-- reg - <int> - I2C slave address
-- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
- DVDD-supply : power supplies for the device as covered in
- Documentation/devicetree/bindings/regulator/regulator.txt
-
-
-Optional properties:
-
-- reset-gpios - GPIO specification for the active low RESET input.
-- ai31xx-micbias-vg - MicBias Voltage setting
- 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
- 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
- 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD
- If this node is not mentioned or if the value is unknown, then
- micbias is set to 2.0V.
-- ai31xx-ocmv - output common-mode voltage setting
- 0 - 1.35V,
- 1 - 1.5V,
- 2 - 1.65V,
- 3 - 1.8V
-
-Deprecated properties:
-
-- gpio-reset - gpio pin number used for codec reset
-
-CODEC output pins:
- * HPL
- * HPR
- * SPL, devices with stereo speaker amp
- * SPR, devices with stereo speaker amp
- * SPK, devices with mono speaker amp
- * MICBIAS
-
-CODEC input pins:
- * MIC1LP, devices with ADC
- * MIC1RP, devices with ADC
- * MIC1LM, devices with ADC
- * AIN1, devices without ADC
- * AIN2, devices without ADC
-
-The pins can be used in referring sound node's audio-routing property.
-
-Example:
-#include <dt-bindings/gpio/gpio.h>
-#include <dt-bindings/sound/tlv320aic31xx.h>
-
-tlv320aic31xx: tlv320aic31xx@18 {
- compatible = "ti,tlv320aic311x";
- reg = <0x18>;
-
- ai31xx-micbias-vg = <MICBIAS_OFF>;
-
- reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
-
- HPVDD-supply = <&regulator>;
- SPRVDD-supply = <&regulator>;
- SPLVDD-supply = <&regulator>;
- AVDD-supply = <&regulator>;
- IOVDD-supply = <&regulator>;
- DVDD-supply = <&regulator>;
-};
diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt
deleted file mode 100644
index 6dfa740e4b2d..000000000000
--- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt
+++ /dev/null
@@ -1,27 +0,0 @@
-Texas Instruments - tpa6130a2 Codec module
-
-The tpa6130a2 serial control bus communicates through I2C protocols
-
-Required properties:
-
-- compatible - "string" - One of:
- "ti,tpa6130a2" - TPA6130A2
- "ti,tpa6140a2" - TPA6140A2
-
-
-- reg - <int> - I2C slave address
-
-- Vdd-supply - <phandle> - power supply regulator
-
-Optional properties:
-
-- power-gpio - gpio pin to power the device
-
-Example:
-
-tpa6130a2: tpa6130a2@60 {
- compatible = "ti,tpa6130a2";
- reg = <0x60>;
- Vdd-supply = <&vmmc2>;
- power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
-};