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authorLinus Torvalds <torvalds@linux-foundation.org>2020-12-15 22:43:47 +0100
committerLinus Torvalds <torvalds@linux-foundation.org>2020-12-15 22:43:47 +0100
commitc367caf1a38b6f0a1aababafd88b00fefa625f9e (patch)
treef622681eff5785d5d15e6b04ca24b15cd7c473f9 /Documentation
parentMerge tag 'net-next-5.11' of git://git.kernel.org/pub/scm/linux/kernel/git/ne... (diff)
parentMerge tag 'asoc-v5.11' of https://git.kernel.org/pub/scm/linux/kernel/git/bro... (diff)
downloadlinux-c367caf1a38b6f0a1aababafd88b00fefa625f9e.tar.xz
linux-c367caf1a38b6f0a1aababafd88b00fefa625f9e.zip
Merge tag 'sound-5.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "Lots of changes (slightly more code increase than usual) at this time, while most of code changes are ASoC driver-specific. Here are some highlights: Core: - The new auxiliary bus implementation for Intel DSP, which will be used by other drivers as well - Lots of ASoC core cleanups and refactoring - UBSAN and KCSAN fixes in rawmidi, sequencer and a few others - Compress-offload API enhancement for the pause during draining HD- and USB-audio: - Enhancements of the USB-audio implicit feedback support, including better full-duplex operations - Continued CA0132 improvements and fixes - A few new quirk entries, HDMI audio fixes ASoC: - Support for boot time selection of Intel DSP firmware, which should help distros/users testing new stuff more easily; the kconfig was moved to boot time option, too - Some basic DPCM support in audio graph card - Removal of old pre-DT Freescale drivers - Support for Allwinner H6 I2S, Analog Devices ADAU1372, Intel Alderlake-S, GMediatek MT8192, NXP i.MX HDMI and XCVR, Realtek RT715, Qualcomm SM8250 and simple GPIO based muxes" * tag 'sound-5.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (445 commits) ALSA: pcm: oss: Fix potential out-of-bounds shift ALSA: usb-audio: Fix potential out-of-bounds shift ALSA: hda/ca0132 - Add ZxR surround DAC setup. ALSA: hda/ca0132 - Add 8051 PLL write helper functions. ALSA: hda/hdmi: packet buffer index must be set before reading value ASoC: SOF: imx: update kernel-doc description ASoC: mediatek: mt8183: delete some unreachable code ASoC: mediatek: mt8183: add PM ops to machine drivers ASoC: topology: Fix wrong size check ASoC: topology: Add missing size check ASoC: SOF: Intel: hda: fix the condition passed to sof_dev_dbg_or_err ASoC: SOF: modify the SOF_DBG flags ASoC: SOF: Intel: hda: remove duplicated status dump ASoC: rt1015p: delay 300ms after SDB pulling high for calibration ASoC: rt1015p: move SDB control from trigger to DAPM ASoC: wm_adsp: remove "ctl" from list on error in wm_adsp_create_control() ALSA: usb-audio: Fix control 'access overflow' errors from chmap ALSA: hda/hdmi: always print pin NIDs as hexadecimal ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions. ...
Diffstat (limited to 'Documentation')
-rw-r--r--Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml7
-rw-r--r--Documentation/devicetree/bindings/sound/adi,adau1372.yaml67
-rw-r--r--Documentation/devicetree/bindings/sound/adi,adau1977.txt61
-rw-r--r--Documentation/devicetree/bindings/sound/adi,adau1977.yaml92
-rw-r--r--Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml6
-rw-r--r--Documentation/devicetree/bindings/sound/audio-graph-card.txt337
-rw-r--r--Documentation/devicetree/bindings/sound/audio-graph-card.yaml57
-rw-r--r--Documentation/devicetree/bindings/sound/audio-graph-port.yaml84
-rw-r--r--Documentation/devicetree/bindings/sound/audio-graph.yaml45
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml66
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,spdif.yaml1
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,xcvr.yaml104
-rw-r--r--Documentation/devicetree/bindings/sound/fsl-asoc-card.txt2
-rw-r--r--Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml138
-rw-r--r--Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml52
-rw-r--r--Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml25
-rw-r--r--Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml44
-rw-r--r--Documentation/devicetree/bindings/sound/nau8315.txt18
-rw-r--r--Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt35
-rw-r--r--Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml112
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml67
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml69
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,sm8250.yaml159
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.txt520
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.yaml447
-rw-r--r--Documentation/devicetree/bindings/sound/rt5682.txt2
-rw-r--r--Documentation/devicetree/bindings/sound/simple-audio-mux.yaml41
-rw-r--r--Documentation/devicetree/bindings/sound/simple-card.yaml6
-rw-r--r--Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt63
-rw-r--r--Documentation/devicetree/bindings/sound/st,stm32-sai.txt107
-rw-r--r--Documentation/devicetree/bindings/sound/st,stm32-sai.yaml200
-rw-r--r--Documentation/sound/alsa-configuration.rst5
32 files changed, 1887 insertions, 1152 deletions
diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml
index d61bc011e820..6f2398cdc82d 100644
--- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml
+++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml
@@ -199,8 +199,6 @@ patternProperties:
description:
From common IIO binding. Used to pipe external sigma delta
modulator or internal ADC output to DFSDM channel.
- This is not required for "st,stm32-dfsdm-pdm" compatibility as
- PDM microphone is binded in Audio DT node.
required:
- io-channels
@@ -235,6 +233,10 @@ patternProperties:
description: child node
properties:
+ compatible:
+ enum:
+ - st,stm32h7-dfsdm-dai
+
"#sound-dai-cells":
const: 0
@@ -244,6 +246,7 @@ patternProperties:
modulator or internal ADC output to DFSDM channel.
required:
+ - compatible
- "#sound-dai-cells"
- io-channels
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1372.yaml b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml
new file mode 100644
index 000000000000..701449311fec
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau1372.yaml
@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau1372.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+
+title: Analog Devices ADAU1372 CODEC
+
+maintainers:
+ - Alexandre Belloni <alexandre.belloni@bootlin.om>
+
+description: |
+ Analog Devices ADAU1372 four inputs and two outputs codec.
+ https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf
+
+properties:
+ compatible:
+ enum:
+ - adi,adau1372
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: "mclk"
+
+ powerdown-gpios:
+ description: GPIO used for hardware power-down.
+ maxItems: 1
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ audio-codec@3c {
+ compatible = "adi,adau1372";
+ reg = <0x3c>;
+ #sound-dai-cells = <0>;
+ clock-names = "mclk";
+ clocks = <&adau1372z_xtal>;
+ };
+ };
+
+ adau1372z_xtal: clock {
+ compatible = "fixed-clock";
+ #clock-cells = <0>;
+ clock-frequency = <12288000>;
+ };
+...
+
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
deleted file mode 100644
index 37f8aad01203..000000000000
--- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt
+++ /dev/null
@@ -1,61 +0,0 @@
-Analog Devices ADAU1977/ADAU1978/ADAU1979
-
-Datasheets:
-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
-
-This driver supports both the I2C and SPI bus.
-
-Required properties:
- - compatible: Should contain one of the following:
- "adi,adau1977"
- "adi,adau1978"
- "adi,adau1979"
-
- - AVDD-supply: analog power supply for the device, please consult
- Documentation/devicetree/bindings/regulator/regulator.txt
-
-Optional properties:
- - reset-gpios: the reset pin for the chip, for more details consult
- Documentation/devicetree/bindings/gpio/gpio.txt
-
- - DVDD-supply: supply voltage for the digital core, please consult
- Documentation/devicetree/bindings/regulator/regulator.txt
-
-- adi,micbias: configures the voltage setting for the MICBIAS pin.
- Select 0/1/2/3/4/5/6/7/8 to specify MICBIAS voltage
- 5V/5.5V/6V/6.5V/7V/7.5V/8V/8.5V/9V
- If not specified the default value will be "7" meaning 8.5 Volts.
- This property is only valid for the ADAU1977
-
-For required properties on SPI, please consult
-Documentation/devicetree/bindings/spi/spi-bus.txt
-
-Required properties on I2C:
-
- - reg: The i2c address. Value depends on the state of ADDR0
- and ADDR1, as wired in hardware.
-
-Examples:
-
- adau1977_spi: adau1977@0 {
- compatible = "adi,adau1977";
- spi-max-frequency = <600000>;
-
- AVDD-supply = <&regulator>;
- DVDD-supply = <&regulator_digital>;
-
- adi,micbias = <3>;
- reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
- };
-
- adau1977_i2c: adau1977@11 {
- compatible = "adi,adau1977";
- reg = <0x11>;
-
- AVDD-supply = <&regulator>;
- DVDD-supply = <&regulator_digital>;
-
- reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
- };
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml
new file mode 100644
index 000000000000..b80454ad97da
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml
@@ -0,0 +1,92 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau1977.yaml#
+$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
+
+ spi-max-frequency: true
+
+ 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
+
+additionalProperties: 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 = <&regulator>;
+ DVDD-supply = <&regulator_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 = <&regulator>;
+ DVDD-supply = <&regulator_digital>;
+
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml
index 112ae00d63c1..a16e37b01e1d 100644
--- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml
+++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml
@@ -24,6 +24,7 @@ properties:
- items:
- const: allwinner,sun50i-a64-i2s
- const: allwinner,sun8i-h3-i2s
+ - const: allwinner,sun50i-h6-i2s
reg:
maxItems: 1
@@ -59,6 +60,7 @@ allOf:
- allwinner,sun8i-a83t-i2s
- allwinner,sun8i-h3-i2s
- allwinner,sun50i-a64-codec-i2s
+ - allwinner,sun50i-h6-i2s
then:
required:
@@ -68,7 +70,9 @@ allOf:
properties:
compatible:
contains:
- const: allwinner,sun8i-a83t-i2s
+ enum:
+ - allwinner,sun8i-a83t-i2s
+ - allwinner,sun8i-h3-i2s
then:
properties:
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt
deleted file mode 100644
index d5f6919a2d69..000000000000
--- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt
+++ /dev/null
@@ -1,337 +0,0 @@
-Audio Graph Card:
-
-Audio Graph Card specifies audio DAI connections of SoC <-> codec.
-It is based on common bindings for device graphs.
-see ${LINUX}/Documentation/devicetree/bindings/graph.txt
-
-Basically, Audio Graph Card property is same as Simple Card.
-see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.yaml
-
-Below are same as Simple-Card.
-
-- label
-- widgets
-- routing
-- dai-format
-- frame-master
-- bitclock-master
-- bitclock-inversion
-- frame-inversion
-- mclk-fs
-- hp-det-gpio
-- mic-det-gpio
-- dai-tdm-slot-num
-- dai-tdm-slot-width
-- clocks / system-clock-frequency
-
-Required properties:
-
-- compatible : "audio-graph-card";
-- dais : list of CPU DAI port{s}
-
-Optional properties:
-- pa-gpios: GPIO used to control external amplifier.
-
------------------------
-Example: Single DAI case
------------------------
-
- sound_card {
- compatible = "audio-graph-card";
-
- dais = <&cpu_port>;
- };
-
- dai-controller {
- ...
- cpu_port: port {
- cpu_endpoint: endpoint {
- remote-endpoint = <&codec_endpoint>;
-
- dai-format = "left_j";
- ...
- };
- };
- };
-
- audio-codec {
- ...
- port {
- codec_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint>;
- };
- };
- };
-
------------------------
-Example: Multi DAI case
------------------------
-
- sound-card {
- compatible = "audio-graph-card";
-
- label = "sound-card";
-
- dais = <&cpu_port0
- &cpu_port1
- &cpu_port2>;
- };
-
- audio-codec@0 {
- ...
- port {
- codec0_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint0>;
- };
- };
- };
-
- audio-codec@1 {
- ...
- port {
- codec1_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint1>;
- };
- };
- };
-
- audio-codec@2 {
- ...
- port {
- codec2_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint2>;
- };
- };
- };
-
- dai-controller {
- ...
- ports {
- cpu_port0: port@0 {
- cpu_endpoint0: endpoint {
- remote-endpoint = <&codec0_endpoint>;
-
- dai-format = "left_j";
- ...
- };
- };
- cpu_port1: port@1 {
- cpu_endpoint1: endpoint {
- remote-endpoint = <&codec1_endpoint>;
-
- dai-format = "i2s";
- ...
- };
- };
- cpu_port2: port@2 {
- cpu_endpoint2: endpoint {
- remote-endpoint = <&codec2_endpoint>;
-
- dai-format = "i2s";
- ...
- };
- };
- };
- };
-
-
------------------------
-Example: Sampling Rate Conversion
------------------------
-
- sound_card {
- compatible = "audio-graph-card";
-
- label = "sound-card";
- prefix = "codec";
- routing = "codec Playback", "DAI0 Playback",
- "DAI0 Capture", "codec Capture";
- convert-rate = <48000>;
-
- dais = <&cpu_port>;
- };
-
- audio-codec {
- ...
- port {
- codec_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint>;
- };
- };
- };
-
- dai-controller {
- ...
- cpu_port: port {
- cpu_endpoint: endpoint {
- remote-endpoint = <&codec_endpoint>;
-
- dai-format = "left_j";
- ...
- };
- };
- };
-
------------------------
-Example: 2 CPU 1 Codec (Mixing)
------------------------
-
- sound_card {
- compatible = "audio-graph-card";
-
- label = "sound-card";
- routing = "codec Playback", "DAI0 Playback",
- "codec Playback", "DAI1 Playback",
- "DAI0 Capture", "codec Capture";
-
- dais = <&cpu_port>;
- };
-
- audio-codec {
- ...
-
- audio-graph-card,prefix = "codec";
- audio-graph-card,convert-rate = <48000>;
- port {
- reg = <0>;
- codec_endpoint0: endpoint@0 {
- remote-endpoint = <&cpu_endpoint0>;
- };
- codec_endpoint1: endpoint@1 {
- remote-endpoint = <&cpu_endpoint1>;
- };
- };
- };
-
- dai-controller {
- ...
- cpu_port: port {
- cpu_endpoint0: endpoint@0 {
- remote-endpoint = <&codec_endpoint0>;
-
- dai-format = "left_j";
- ...
- };
- cpu_endpoint1: endpoint@1 {
- remote-endpoint = <&codec_endpoint1>;
-
- dai-format = "left_j";
- ...
- };
- };
- };
-
------------------------
-Example: Multi DAI with DPCM
------------------------
-
- CPU0 ------ ak4613
- CPU1 ------ HDMI
- CPU2 ------ PCM3168A-p /* DPCM 1ch/2ch */
- CPU3 --/ /* DPCM 3ch/4ch */
- CPU4 --/ /* DPCM 5ch/6ch */
- CPU5 --/ /* DPCM 7ch/8ch */
- CPU6 ------ PCM3168A-c
-
- sound_card: sound {
- compatible = "audio-graph-card";
-
- label = "sound-card";
-
- routing = "pcm3168a Playback", "DAI2 Playback",
- "pcm3168a Playback", "DAI3 Playback",
- "pcm3168a Playback", "DAI4 Playback",
- "pcm3168a Playback", "DAI5 Playback";
-
- dais = <&snd_port0 /* ak4613 */
- &snd_port1 /* HDMI0 */
- &snd_port2 /* pcm3168a playback */
- &snd_port3 /* pcm3168a capture */
- >;
- };
-
- ak4613: codec@10 {
- ...
- port {
- ak4613_endpoint: endpoint {
- remote-endpoint = <&rsnd_endpoint0>;
- };
- };
- };
-
- pcm3168a: audio-codec@44 {
- ...
- audio-graph-card,prefix = "pcm3168a";
- audio-graph-card,convert-channels = <8>; /* TDM Split */
- ports {
- port@0 {
- reg = <0>;
- pcm3168a_endpoint_p1: endpoint@1 {
- remote-endpoint = <&rsnd_endpoint2>;
- ...
- };
- pcm3168a_endpoint_p2: endpoint@2 {
- remote-endpoint = <&rsnd_endpoint3>;
- ...
- };
- pcm3168a_endpoint_p3: endpoint@3 {
- remote-endpoint = <&rsnd_endpoint4>;
- ...
- };
- pcm3168a_endpoint_p4: endpoint@4 {
- remote-endpoint = <&rsnd_endpoint5>;
- ...
- };
- };
- port@1 {
- reg = <1>;
- pcm3168a_endpoint_c: endpoint {
- remote-endpoint = <&rsnd_endpoint6>;
- ...
- };
- };
- };
- };
-
- &sound {
- ports {
- snd_port0: port@0 {
- rsnd_endpoint0: endpoint {
- remote-endpoint = <&ak4613_endpoint>;
- ...
- };
- };
- snd_port1: port@1 {
- rsnd_endpoint1: endpoint {
- remote-endpoint = <&dw_hdmi0_snd_in>;
- ...
- };
- };
- snd_port2: port@2 {
- #address-cells = <1>;
- #size-cells = <0>;
- rsnd_endpoint2: endpoint@2 {
- remote-endpoint = <&pcm3168a_endpoint_p1>;
- ...
- };
- rsnd_endpoint3: endpoint@3 {
- remote-endpoint = <&pcm3168a_endpoint_p2>;
- ...
- };
- rsnd_endpoint4: endpoint@4 {
- remote-endpoint = <&pcm3168a_endpoint_p3>;
- ...
- };
- rsnd_endpoint5: endpoint@5 {
- remote-endpoint = <&pcm3168a_endpoint_p4>;
- ...
- };
- };
- snd_port3: port@6 {
- rsnd_endpoint6: endpoint {
- remote-endpoint = <&pcm3168a_endpoint_c>;
- ...
- };
- };
- };
- };
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml
new file mode 100644
index 000000000000..109e55f9e597
--- /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 Device Tree Bindings
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+allOf:
+ - $ref: /schemas/sound/audio-graph.yaml#
+
+properties:
+ compatible:
+ enum:
+ - audio-graph-card
+ - audio-graph-scu-card
+
+required:
+ - compatible
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound {
+ compatible = "audio-graph-card";
+
+ dais = <&cpu_port_a>;
+ };
+
+ cpu {
+ /*
+ * dai-controller own settings
+ */
+
+ port {
+ cpu_endpoint: endpoint {
+ remote-endpoint = <&codec_endpoint>;
+ dai-format = "left_j";
+ };
+ };
+ };
+
+ codec {
+ /*
+ * codec own settings
+ */
+
+ port {
+ codec_endpoint: endpoint {
+ remote-endpoint = <&cpu_endpoint>;
+ };
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml
new file mode 100644
index 000000000000..2005014161be
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml
@@ -0,0 +1,84 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/audio-graph-port.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Graph Card 'port' Node Bindings
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+select: false
+
+properties:
+ port:
+ description: single OF-Graph subnode
+ type: object
+ properties:
+ reg:
+ maxItems: 1
+ prefix:
+ description: "device name prefix"
+ $ref: /schemas/types.yaml#/definitions/string
+ 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
+ patternProperties:
+ "^endpoint(@[0-9a-f]+)?":
+ type: object
+ properties:
+ remote-endpoint:
+ maxItems: 1
+ mclk-fs:
+ description: |
+ Multiplication factor between stream rate and codec mclk.
+ When defined, mclk-fs property defined in dai-link sub nodes are
+ ignored.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ frame-inversion:
+ description: dai-link uses frame clock inversion
+ $ref: /schemas/types.yaml#/definitions/flag
+ bitclock-inversion:
+ description: dai-link uses bit clock inversion
+ $ref: /schemas/types.yaml#/definitions/flag
+ frame-master:
+ description: Indicates dai-link frame master.
+ $ref: /schemas/types.yaml#/definitions/phandle
+ bitclock-master:
+ description: Indicates dai-link bit clock master
+ $ref: /schemas/types.yaml#/definitions/phandle
+ dai-format:
+ description: audio format.
+ items:
+ enum:
+ - i2s
+ - right_j
+ - left_j
+ - dsp_a
+ - dsp_b
+ - ac97
+ - pdm
+ - msb
+ - lsb
+ convert-rate:
+ 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
+
+ required:
+ - remote-endpoint
+
+ ports:
+ description: multi OF-Graph subnode
+ type: object
+ patternProperties:
+ "^port(@[0-9a-f]+)?":
+ $ref: "#/properties/port"
+
+additionalProperties: true
diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml
new file mode 100644
index 000000000000..4b46794e5153
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml
@@ -0,0 +1,45 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/audio-graph.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Graph Device Tree Bindings
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+ dais:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ label:
+ maxItems: 1
+ prefix:
+ description: "device name prefix"
+ $ref: /schemas/types.yaml#/definitions/string
+ routing:
+ description: |
+ A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's source.
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ widgets:
+ description: User specified audio sound widgets.
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ convert-rate:
+ description: CPU to Codec rate convert.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ convert-channels:
+ description: CPU to Codec rate channels.
+ $ref: /schemas/types.yaml#/definitions/uint32
+ pa-gpios:
+ maxItems: 1
+ hp-det-gpio:
+ maxItems: 1
+ mic-det-gpio:
+ maxItems: 1
+
+required:
+ - dais
+
+additionalProperties: true
diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml
new file mode 100644
index 000000000000..aa4be7170717
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml
@@ -0,0 +1,66 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio Subsystem to HDMI RTX Subsystem Controller
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+ compatible:
+ const: fsl,imx8mp-aud2htx
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Peripheral clock
+
+ clock-names:
+ items:
+ - const: bus
+
+ dmas:
+ items:
+ - description: DMA controller phandle and request line for TX
+
+ dma-names:
+ items:
+ - const: tx
+
+ power-domains:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/imx8mp-clock.h>
+
+ aud2htx: aud2htx@30cb0000 {
+ compatible = "fsl,imx8mp-aud2htx";
+ reg = <0x30cb0000 0x10000>;
+ interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>;
+ clock-names = "bus";
+ dmas = <&sdma2 26 2 0>;
+ dma-names = "tx";
+ power-domains = <&audiomix_pd>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
index 2ac671f5cb9b..50449b6d1048 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
@@ -20,6 +20,7 @@ properties:
- fsl,imx35-spdif
- fsl,vf610-spdif
- fsl,imx6sx-spdif
+ - fsl,imx8qm-spdif
reg:
maxItems: 1
diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml
new file mode 100644
index 000000000000..223b8ea693dc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml
@@ -0,0 +1,104 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,xcvr.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio Transceiver (XCVR) Controller
+
+maintainers:
+ - Viorel Suman <viorel.suman@nxp.com>
+
+description: |
+ NXP XCVR (Audio Transceiver) is a on-chip functional module
+ that allows CPU to receive and transmit digital audio via
+ HDMI2.1 eARC, HDMI1.4 ARC and SPDIF.
+
+properties:
+ $nodename:
+ pattern: "^xcvr@.*"
+
+ compatible:
+ enum:
+ - fsl,imx8mp-xcvr
+
+ reg:
+ items:
+ - description: 20K RAM for code and data
+ - description: registers space
+ - description: RX FIFO address
+ - description: TX FIFO address
+
+ reg-names:
+ items:
+ - const: ram
+ - const: regs
+ - const: rxfifo
+ - const: txfifo
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: Peripheral clock
+ - description: PHY clock
+ - description: SPBA clock
+ - description: PLL clock
+
+ clock-names:
+ items:
+ - const: ipg
+ - const: phy
+ - const: spba
+ - const: pll_ipg
+
+ dmas:
+ items:
+ - description: DMA controller phandle and request line for RX
+ - description: DMA controller phandle and request line for TX
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+ resets:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - reg-names
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - resets
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/imx8mp-clock.h>
+ #include <dt-bindings/reset/imx8mp-reset.h>
+
+ xcvr: xcvr@30cc0000 {
+ compatible = "fsl,imx8mp-xcvr";
+ reg = <0x30cc0000 0x800>,
+ <0x30cc0800 0x400>,
+ <0x30cc0c00 0x080>,
+ <0x30cc0e00 0x080>;
+ reg-names = "ram", "regs", "rxfifo", "txfifo";
+ interrupts = <0x0 128 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_IPG>,
+ <&audiomix_clk IMX8MP_CLK_AUDIOMIX_EARC_PHY>,
+ <&audiomix_clk IMX8MP_CLK_AUDIOMIX_SPBA2_ROOT>,
+ <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUDPLL_ROOT>;
+ clock-names = "ipg", "phy", "spba", "pll_ipg";
+ dmas = <&sdma2 30 2 0>, <&sdma2 31 2 0>;
+ dma-names = "rx", "tx";
+ resets = <&audiomix_reset 0>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index f339be62e7e4..90d9e9d81624 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -40,6 +40,8 @@ The compatible list for this generic sound card currently:
"fsl,imx-audio-tlv320aic32x4"
+ "fsl,imx-audio-si476x"
+
Required properties:
- compatible : Contains one of entries in the compatible list.
diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml
new file mode 100644
index 000000000000..5095b780e2c7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.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/google,sc7180-trogdor.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Google SC7180-Trogdor ASoC sound card driver
+
+maintainers:
+ - Rohit kumar <rohitkr@codeaurora.org>
+ - Cheng-Yi Chiang <cychiang@chromium.org>
+
+description:
+ This binding describes the SC7180 sound card which uses LPASS for audio.
+
+properties:
+ compatible:
+ enum:
+ - google,sc7180-trogdor
+ - google,sc7180-coachz
+
+ audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 0
+
+ dmic-gpios:
+ maxItems: 1
+ description: GPIO for switching between DMICs
+
+patternProperties:
+ "^dai-link(@[0-9])?$":
+ description:
+ Each subnode represents a dai link. Subnodes of each dai links would be
+ cpu/codec dais.
+
+ type: object
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name.
+ $ref: /schemas/types.yaml#/definitions/string
+ maxItems: 1
+
+ reg:
+ description: dai link address.
+
+ cpu:
+ description: Holds subnode which indicates cpu dai.
+ type: object
+ properties:
+ sound-dai: true
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ properties:
+ sound-dai: true
+
+ required:
+ - link-name
+ - cpu
+ - codec
+
+ additionalProperties: false
+
+required:
+ - compatible
+ - model
+ - "#address-cells"
+ - "#size-cells"
+
+additionalProperties: false
+
+examples:
+
+ - |
+ sound {
+ compatible = "google,sc7180-trogdor";
+ model = "sc7180-rt5682-max98357a-2mic";
+
+ audio-routing =
+ "Headphone Jack", "HPOL",
+ "Headphone Jack", "HPOR";
+
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ dmic-gpios = <&tlmm 86 0>;
+
+ dai-link@0 {
+ link-name = "MultiMedia0";
+ reg = <0>;
+ cpu {
+ sound-dai = <&lpass_cpu 0>;
+ };
+
+ codec {
+ sound-dai = <&alc5682 0>;
+ };
+ };
+
+ dai-link@1 {
+ link-name = "MultiMedia1";
+ reg = <1>;
+ cpu {
+ sound-dai = <&lpass_cpu 1>;
+ };
+
+ codec {
+ sound-dai = <&max98357a>;
+ };
+ };
+
+ dai-link@2 {
+ link-name = "MultiMedia2";
+ reg = <2>;
+ cpu {
+ sound-dai = <&lpass_hdmi 0>;
+ };
+
+ codec {
+ sound-dai = <&msm_dp>;
+ };
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml
new file mode 100644
index 000000000000..d5474f83ac2c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml
@@ -0,0 +1,52 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/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:
+ description: The phandle of an CPU DAI controller
+
+ hdmi-out:
+ 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:
+ 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/marvell,mmp-sspa.yaml b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml
index 6d20a24a2ae9..234f64a32184 100644
--- a/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml
+++ b/Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml
@@ -9,6 +9,9 @@ title: Marvel SSPA Digital Audio Interface Bindings
maintainers:
- Lubomir Rintel <lkundrak@v3.sk>
+allOf:
+ - $ref: audio-graph-port.yaml#
+
properties:
$nodename:
pattern: "^audio-controller(@.*)?$"
@@ -58,29 +61,9 @@ properties:
type: object
properties:
- remote-endpoint: true
-
- frame-master:
- type: boolean
- description: SoC generates the frame clock
-
- bitclock-master:
- type: boolean
- description: SoC generates the bit clock
-
dai-format:
- $ref: /schemas/types.yaml#/definitions/string
- description: The digital audio format
const: i2s
- required:
- - remote-endpoint
-
- required:
- - endpoint
-
- additionalProperties: false
-
required:
- "#sound-dai-cells"
- compatible
@@ -112,8 +95,6 @@ examples:
port {
endpoint {
remote-endpoint = <&rt5631_0>;
- frame-master;
- bitclock-master;
dai-format = "i2s";
};
};
diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
new file mode 100644
index 000000000000..bf8c8ba25009
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
@@ -0,0 +1,44 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mt8192-mt6359-rt1015-rt5682.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Mediatek MT8192 with MT6359, RT1015 and RT5682 ASoC sound card driver
+
+maintainers:
+ - Jiaxin Yu <jiaxin.yu@mediatek.com>
+ - Shane Chien <shane.chien@mediatek.com>
+
+description:
+ This binding describes the MT8192 sound card.
+
+properties:
+ compatible:
+ enum:
+ - mediatek,mt8192_mt6359_rt1015_rt5682
+ - mediatek,mt8192_mt6359_rt1015p_rt5682
+
+ mediatek,platform:
+ $ref: "/schemas/types.yaml#/definitions/phandle"
+ description: The phandle of MT8192 ASoC platform.
+
+additionalProperties: false
+
+required:
+ - compatible
+ - mediatek,platform
+
+examples:
+ - |
+
+ sound: mt8192-sound {
+ compatible = "mediatek,mt8192_mt6359_rt1015_rt5682";
+ mediatek,platform = <&afe>;
+ pinctrl-names = "aud_clk_mosi_off",
+ "aud_clk_mosi_on";
+ pinctrl-0 = <&aud_clk_mosi_off>;
+ pinctrl-1 = <&aud_clk_mosi_on>;
+ };
+
+...
diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt
new file mode 100644
index 000000000000..6eaec46f384c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8315.txt
@@ -0,0 +1,18 @@
+Nuvoton NAU8315 Mono Class-D Amplifier
+
+Required properties:
+- compatible : "nuvoton,nau8315"
+
+Optional properties:
+- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin.
+ If this option is not specified then driver does not manage
+ the pin state (e.g. chip is always on).
+
+Example:
+
+#include <dt-bindings/gpio/gpio.h>
+
+nau8315 {
+ compatible = "nuvoton,nau8315";
+ enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
deleted file mode 100644
index 21cd310963b1..000000000000
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
+++ /dev/null
@@ -1,35 +0,0 @@
-NVIDIA Tegra30 HDA controller
-
-Required properties:
-- compatible : For Tegra30, must contain "nvidia,tegra30-hda". Otherwise,
- must contain '"nvidia,<chip>-hda", "nvidia,tegra30-hda"', where <chip> is
- tegra114, tegra124, or tegra132.
-- reg : Should contain the HDA registers location and length.
-- interrupts : The interrupt from the HDA controller.
-- clocks : Must contain an entry for each required entry in clock-names.
- See ../clocks/clock-bindings.txt for details.
-- clock-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x
-- resets : Must contain an entry for each entry in reset-names.
- See ../reset/reset.txt for details.
-- reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x
-
-Optional properties:
-- nvidia,model : The user-visible name of this sound complex. Since the property
- is optional, legacy boards can use default name provided in hda driver.
-
-Example:
-
-hda@70030000 {
- compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda";
- reg = <0x0 0x70030000 0x0 0x10000>;
- interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
- clocks = <&tegra_car TEGRA124_CLK_HDA>,
- <&tegra_car TEGRA124_CLK_HDA2HDMI>,
- <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>;
- clock-names = "hda", "hda2hdmi", "hda2codec_2x";
- resets = <&tegra_car 125>, /* hda */
- <&tegra_car 128>, /* hda2hdmi */
- <&tegra_car 111>; /* hda2codec_2x */
- reset-names = "hda", "hda2hdmi", "hda2codec_2x";
- nvidia,model = "jetson-tk1-hda";
-};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml
new file mode 100644
index 000000000000..e543a6123792
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.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/nvidia,tegra30-hda.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra HDA controller
+
+description: |
+ The High Definition Audio (HDA) block provides a serial interface to
+ audio codec. It supports multiple input and output streams.
+
+maintainers:
+ - Thierry Reding <treding@nvidia.com>
+ - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+ $nodename:
+ pattern: "^hda@[0-9a-f]*$"
+
+ compatible:
+ oneOf:
+ - const: nvidia,tegra30-hda
+ - items:
+ - enum:
+ - nvidia,tegra194-hda
+ - nvidia,tegra186-hda
+ - nvidia,tegra210-hda
+ - nvidia,tegra124-hda
+ - const: nvidia,tegra30-hda
+ - items:
+ - const: nvidia,tegra132-hda
+ - const: nvidia,tegra124-hda
+ - const: nvidia,tegra30-hda
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ description: The interrupt from the HDA controller
+ maxItems: 1
+
+ clocks:
+ maxItems: 3
+
+ clock-names:
+ maxItems: 3
+ items:
+ - const: hda
+ - const: hda2hdmi
+ - const: hda2codec_2x
+
+ resets:
+ maxItems: 3
+
+ reset-names:
+ maxItems: 3
+ items:
+ - const: hda
+ - const: hda2hdmi
+ - const: hda2codec_2x
+
+ power-domains:
+ maxItems: 1
+
+ interconnects:
+ maxItems: 2
+
+ interconnect-names:
+ items:
+ - const: dma-mem
+ - const: write
+
+ iommus:
+ maxItems: 1
+
+ nvidia,model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: |
+ The user-visible name of this sound complex. If this property is
+ not specified then boards can use default name provided in hda driver.
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include<dt-bindings/clock/tegra124-car-common.h>
+ #include<dt-bindings/interrupt-controller/arm-gic.h>
+
+ hda@70030000 {
+ compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda";
+ reg = <0x70030000 0x10000>;
+ interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&tegra_car TEGRA124_CLK_HDA>,
+ <&tegra_car TEGRA124_CLK_HDA2HDMI>,
+ <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>;
+ clock-names = "hda", "hda2hdmi", "hda2codec_2x";
+ resets = <&tegra_car 125>, /* hda */
+ <&tegra_car 128>, /* hda2hdmi */
+ <&tegra_car 111>; /* hda2codec_2x */
+ reset-names = "hda", "hda2hdmi", "hda2codec_2x";
+ nvidia,model = "jetson-tk1-hda";
+ };
+
+...
diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml
new file mode 100644
index 000000000000..679b49cbe30f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml
@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) VA Macro audio codec DT bindings
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+ compatible:
+ const: qcom,sm8250-lpass-va-macro
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ clocks:
+ maxItems: 3
+
+ clock-names:
+ items:
+ - const: mclk
+ - const: core
+ - const: dcodec
+
+ clock-output-names:
+ items:
+ - const: fsgen
+
+ qcom,dmic-sample-rate:
+ description: dmic sample rate
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ vdd-micb-supply:
+ description: phandle to voltage regulator of MIC Bias
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ codec@3370000 {
+ compatible = "qcom,sm8250-lpass-va-macro";
+ reg = <0x3370000 0x1000>;
+ #sound-dai-cells = <1>;
+ #clock-cells = <0>;
+ clocks = <&aoncc 0>,
+ <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>;
+ clock-names = "mclk", "core", "dcodec";
+ clock-output-names = "fsgen";
+ qcom,dmic-sample-rate = <600000>;
+ vdd-micb-supply = <&vreg_s4a_1p8>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml
new file mode 100644
index 000000000000..435b019a1e3d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml
@@ -0,0 +1,69 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-wsa-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) VA Macro audio codec DT bindings
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+ compatible:
+ const: qcom,sm8250-lpass-wsa-macro
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ clocks:
+ maxItems: 5
+
+ clock-names:
+ items:
+ - const: mclk
+ - const: npl
+ - const: macro
+ - const: dcodec
+ - const: fsgen
+
+ clock-output-names:
+ items:
+ - const: mclk
+
+ qcom,dmic-sample-rate:
+ description: dmic sample rate
+ $ref: /schemas/types.yaml#/definitions/uint32
+
+ vdd-micb-supply:
+ description: phandle to voltage regulator of MIC Bias
+
+required:
+ - compatible
+ - reg
+ - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/sound/qcom,q6afe.h>
+ codec@3240000 {
+ compatible = "qcom,sm8250-lpass-wsa-macro";
+ reg = <0x3240000 0x1000>;
+ #sound-dai-cells = <1>;
+ #clock-cells = <0>;
+ clocks = <&audiocc 1>,
+ <&audiocc 0>,
+ <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+ <&vamacro>;
+ clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
+ clock-output-names = "mclk";
+ };
diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
new file mode 100644
index 000000000000..72ad9ab91832
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
@@ -0,0 +1,159 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,sm8250.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Technologies Inc. SM8250 ASoC sound card driver
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+ This bindings describes SC8250 SoC based sound cards
+ which uses LPASS internal codec for audio.
+
+properties:
+ compatible:
+ oneOf:
+ - const: qcom,sm8250-sndcard
+ - const: qcom,qrb5165-rb5-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.
+
+ 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
+ properties:
+ sound-dai: true
+
+ platform:
+ description: Holds subnode which indicates platform dai.
+ type: object
+ properties:
+ sound-dai: true
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ properties:
+ sound-dai: true
+
+ 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",
+ "MM_DL1", "MultiMedia1 Playback",
+ "MM_DL2", "MultiMedia2 Playback",
+ "MultiMedia3 Capture", "MM_UL3";
+
+ 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 = <&lt9611_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>;
+ };
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index b39743d3f7c4..b731f16aea84 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -253,523 +253,3 @@ This is example of TDM 6ch.
Driver can automatically switches TDM <-> stereo mode in this case.
see "Example: simple sound card for TDM"
-
-=============================================
-Required properties:
-=============================================
-
-- compatible : "renesas,rcar_sound-<soctype>", fallbacks
- "renesas,rcar_sound-gen1" if generation1, and
- "renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
- "renesas,rcar_sound-gen3" if generation3 (or RZ/G2)
- Examples with soctypes are:
- - "renesas,rcar_sound-r8a7742" (RZ/G1H)
- - "renesas,rcar_sound-r8a7743" (RZ/G1M)
- - "renesas,rcar_sound-r8a7744" (RZ/G1N)
- - "renesas,rcar_sound-r8a7745" (RZ/G1E)
- - "renesas,rcar_sound-r8a77470" (RZ/G1C)
- - "renesas,rcar_sound-r8a774a1" (RZ/G2M)
- - "renesas,rcar_sound-r8a774b1" (RZ/G2N)
- - "renesas,rcar_sound-r8a774c0" (RZ/G2E)
- - "renesas,rcar_sound-r8a774e1" (RZ/G2H)
- - "renesas,rcar_sound-r8a7778" (R-Car M1A)
- - "renesas,rcar_sound-r8a7779" (R-Car H1)
- - "renesas,rcar_sound-r8a7790" (R-Car H2)
- - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
- - "renesas,rcar_sound-r8a7793" (R-Car M2-N)
- - "renesas,rcar_sound-r8a7794" (R-Car E2)
- - "renesas,rcar_sound-r8a7795" (R-Car H3)
- - "renesas,rcar_sound-r8a7796" (R-Car M3-W)
- - "renesas,rcar_sound-r8a77965" (R-Car M3-N)
- - "renesas,rcar_sound-r8a77990" (R-Car E3)
- - "renesas,rcar_sound-r8a77995" (R-Car D3)
-- reg : Should contain the register physical address.
- required register is
- SRU/ADG/SSI if generation1
- SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3
- Select extended AUDIO-DMAC-periperi address if SoC has it,
- otherwise select normal AUDIO-DMAC-periperi address.
-- reg-names : Should contain the register names.
- scu/adg/ssi if generation1
- scu/adg/ssiu/ssi/audmapp if generation2/generation3
-- rcar_sound,ssi : Should contain SSI feature.
- The number of SSI subnode should be same as HW.
- see below for detail.
-- rcar_sound,ssiu : Should contain SSIU feature.
- The number of SSIU subnode should be same as HW.
- see below for detail.
-- rcar_sound,src : Should contain SRC feature.
- The number of SRC subnode should be same as HW.
- see below for detail.
-- rcar_sound,ctu : Should contain CTU feature.
- The number of CTU subnode should be same as HW.
- see below for detail.
-- rcar_sound,mix : Should contain MIX feature.
- The number of MIX subnode should be same as HW.
- see below for detail.
-- rcar_sound,dvc : Should contain DVC feature.
- The number of DVC subnode should be same as HW.
- see below for detail.
-- rcar_sound,dai : DAI contents.
- The number of DAI subnode should be same as HW.
- see below for detail.
-- #sound-dai-cells : it must be 0 if your system is using single DAI
- it must be 1 if your system is using multi DAI
-- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks.
-- clock-names : List of necessary clock names.
- "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X",
- "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i"
-
-Optional properties:
-- #clock-cells : it must be 0 if your system has audio_clkout
- it must be 1 if your system has audio_clkout0/1/2/3
-- clock-frequency : for all audio_clkout0/1/2/3
-- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn
- is asynchronizes with lr-clock.
-- resets : References to SSI resets.
-- reset-names : List of valid reset names.
- "ssi-all", "ssi.X"
-
-SSI subnode properties:
-- interrupts : Should contain SSI interrupt for PIO transfer
-- shared-pin : if shared clock pin
-- pio-transfer : use PIO transfer mode
-- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case
-- dma : Should contain Audio DMAC entry
-- dma-names : SSI case "rx" (=playback), "tx" (=capture)
- Deprecated: see SSIU subnode properties
- SSIU case "rxu" (=playback), "txu" (=capture)
-
-SSIU subnode properties:
-- dma : Should contain Audio DMAC entry
-- dma-names : "rx" (=playback), "tx" (=capture)
-
-SRC subnode properties:
-- dma : Should contain Audio DMAC entry
-- dma-names : "rx" (=playback), "tx" (=capture)
-
-DVC subnode properties:
-- dma : Should contain Audio DMAC entry
-- dma-names : "tx" (=playback/capture)
-
-DAI subnode properties:
-- playback : list of playback modules
-- capture : list of capture modules
-
-
-=============================================
-Example:
-=============================================
-
-rcar_sound: sound@ec500000 {
- #sound-dai-cells = <1>;
- compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2";
- reg = <0 0xec500000 0 0x1000>, /* SCU */
- <0 0xec5a0000 0 0x100>, /* ADG */
- <0 0xec540000 0 0x1000>, /* SSIU */
- <0 0xec541000 0 0x1280>, /* SSI */
- <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/
- reg-names = "scu", "adg", "ssiu", "ssi", "audmapp";
-
- clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>,
- <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>,
- <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>,
- <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>,
- <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>,
- <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>,
- <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>,
- <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>,
- <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>,
- <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>,
- <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>,
- <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>,
- <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>;
- clock-names = "ssi-all",
- "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5",
- "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0",
- "src.9", "src.8", "src.7", "src.6", "src.5",
- "src.4", "src.3", "src.2", "src.1", "src.0",
- "dvc.0", "dvc.1",
- "clk_a", "clk_b", "clk_c", "clk_i";
-
- rcar_sound,dvc {
- dvc0: dvc-0 {
- dmas = <&audma0 0xbc>;
- dma-names = "tx";
- };
- dvc1: dvc-1 {
- dmas = <&audma0 0xbe>;
- dma-names = "tx";
- };
- };
-
- rcar_sound,mix {
- mix0: mix-0 { };
- mix1: mix-1 { };
- };
-
- rcar_sound,ctu {
- ctu00: ctu-0 { };
- ctu01: ctu-1 { };
- ctu02: ctu-2 { };
- ctu03: ctu-3 { };
- ctu10: ctu-4 { };
- ctu11: ctu-5 { };
- ctu12: ctu-6 { };
- ctu13: ctu-7 { };
- };
-
- rcar_sound,src {
- src0: src-0 {
- interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x85>, <&audma1 0x9a>;
- dma-names = "rx", "tx";
- };
- src1: src-1 {
- interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x87>, <&audma1 0x9c>;
- dma-names = "rx", "tx";
- };
- src2: src-2 {
- interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x89>, <&audma1 0x9e>;
- dma-names = "rx", "tx";
- };
- src3: src-3 {
- interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x8b>, <&audma1 0xa0>;
- dma-names = "rx", "tx";
- };
- src4: src-4 {
- interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x8d>, <&audma1 0xb0>;
- dma-names = "rx", "tx";
- };
- src5: src-5 {
- interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x8f>, <&audma1 0xb2>;
- dma-names = "rx", "tx";
- };
- src6: src-6 {
- interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x91>, <&audma1 0xb4>;
- dma-names = "rx", "tx";
- };
- src7: src-7 {
- interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x93>, <&audma1 0xb6>;
- dma-names = "rx", "tx";
- };
- src8: src-8 {
- interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x95>, <&audma1 0xb8>;
- dma-names = "rx", "tx";
- };
- src9: src-9 {
- interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x97>, <&audma1 0xba>;
- dma-names = "rx", "tx";
- };
- };
-
- rcar_sound,ssiu {
- ssiu00: ssiu-0 {
- dmas = <&audma0 0x15>, <&audma1 0x16>;
- dma-names = "rx", "tx";
- };
- ssiu01: ssiu-1 {
- dmas = <&audma0 0x35>, <&audma1 0x36>;
- dma-names = "rx", "tx";
- };
-
- ...
-
- ssiu95: ssiu-49 {
- dmas = <&audma0 0xA5>, <&audma1 0xA6>;
- dma-names = "rx", "tx";
- };
- ssiu96: ssiu-50 {
- dmas = <&audma0 0xA7>, <&audma1 0xA8>;
- dma-names = "rx", "tx";
- };
- ssiu97: ssiu-51 {
- dmas = <&audma0 0xA9>, <&audma1 0xAA>;
- dma-names = "rx", "tx";
- };
- };
-
- rcar_sound,ssi {
- ssi0: ssi-0 {
- interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x01>, <&audma1 0x02>;
- dma-names = "rx", "tx";
- };
- ssi1: ssi-1 {
- interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x03>, <&audma1 0x04>;
- dma-names = "rx", "tx";
- };
-
- ...
-
- ssi8: ssi-8 {
- interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x11>, <&audma1 0x12>;
- dma-names = "rx", "tx";
- };
- ssi9: ssi-9 {
- interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&audma0 0x13>, <&audma1 0x14>;
- dma-names = "rx", "tx";
- };
- };
-
- rcar_sound,dai {
- dai0 {
- playback = <&ssi5 &src5>;
- capture = <&ssi6>;
- };
- dai1 {
- playback = <&ssi3>;
- };
- dai2 {
- capture = <&ssi4>;
- };
- dai3 {
- playback = <&ssi7>;
- };
- dai4 {
- capture = <&ssi8>;
- };
- };
-};
-
-=============================================
-Example: simple sound card
-=============================================
-
- rsnd_ak4643: sound {
- compatible = "simple-audio-card";
-
- simple-audio-card,format = "left_j";
- simple-audio-card,bitclock-master = <&sndcodec>;
- simple-audio-card,frame-master = <&sndcodec>;
-
- sndcpu: simple-audio-card,cpu {
- sound-dai = <&rcar_sound>;
- };
-
- sndcodec: simple-audio-card,codec {
- sound-dai = <&ak4643>;
- clocks = <&audio_clock>;
- };
- };
-
-&rcar_sound {
- pinctrl-0 = <&sound_pins &sound_clk_pins>;
- pinctrl-names = "default";
-
- /* Single DAI */
- #sound-dai-cells = <0>;
-
-
- rcar_sound,dai {
- dai0 {
- playback = <&ssi0 &src2 &dvc0>;
- capture = <&ssi1 &src3 &dvc1>;
- };
- };
-};
-
-&ssi1 {
- shared-pin;
-};
-
-=============================================
-Example: simple sound card for Asynchronous mode
-=============================================
-
-sound {
- compatible = "simple-scu-audio-card";
- ...
- /*
- * SRC Asynchronous mode setting
- * Playback:
- * All input data will be converted to 48kHz
- * Capture:
- * Inputed 48kHz data will be converted to
- * system specified Hz
- */
- simple-audio-card,convert-rate = <48000>;
- ...
- simple-audio-card,cpu {
- sound-dai = <&rcar_sound>;
- };
- simple-audio-card,codec {
- ...
- };
-};
-
-=============================================
-Example: simple sound card for channel convert
-=============================================
-
-sound {
- compatible = "simple-scu-audio-card";
- ...
- /*
- * CTU setting
- * All input data will be converted to 2ch
- * as output data
- */
- simple-audio-card,convert-channels = <2>;
- ...
- simple-audio-card,cpu {
- sound-dai = <&rcar_sound>;
- };
- simple-audio-card,codec {
- ...
- };
-};
-
-=============================================
-Example: simple sound card for MIXer
-=============================================
-
-sound {
- compatible = "simple-scu-audio-card";
- ...
- simple-audio-card,cpu@0 {
- sound-dai = <&rcar_sound 0>;
- };
- simple-audio-card,cpu@1 {
- sound-dai = <&rcar_sound 1>;
- };
- simple-audio-card,codec {
- ...
- };
-};
-
-&rcar_sound {
- ...
- rcar_sound,dai {
- dai0 {
- playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>;
- };
- dai1 {
- playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>;
- };
- };
-};
-
-=============================================
-Example: simple sound card for TDM
-=============================================
-
-rsnd_tdm: sound {
- compatible = "simple-audio-card";
-
- simple-audio-card,format = "left_j";
- simple-audio-card,bitclock-master = <&sndcodec>;
- simple-audio-card,frame-master = <&sndcodec>;
-
- sndcpu: simple-audio-card,cpu {
- sound-dai = <&rcar_sound>;
- dai-tdm-slot-num = <6>;
- };
-
- sndcodec: simple-audio-card,codec {
- sound-dai = <&xxx>;
- };
-};
-
-=============================================
-Example: simple sound card for TDM Split
-=============================================
-
-sound_card: sound {
- compatible = "audio-graph-scu-card";
- prefix = "xxxx";
- routing = "xxxx Playback", "DAI0 Playback",
- "xxxx Playback", "DAI1 Playback",
- "xxxx Playback", "DAI2 Playback",
- "xxxx Playback", "DAI3 Playback";
- convert-channels = <8>; /* TDM Split */
-
- dais = <&rsnd_port0 /* playback ch1/ch2 */
- &rsnd_port1 /* playback ch3/ch4 */
- &rsnd_port2 /* playback ch5/ch6 */
- &rsnd_port3 /* playback ch7/ch8 */
- >;
-};
-
-audio-codec {
- ...
- port {
- codec_0: endpoint@1 {
- remote-endpoint = <&rsnd_ep0>;
- };
- codec_1: endpoint@2 {
- remote-endpoint = <&rsnd_ep1>;
- };
- codec_2: endpoint@3 {
- remote-endpoint = <&rsnd_ep2>;
- };
- codec_3: endpoint@4 {
- remote-endpoint = <&rsnd_ep3>;
- };
- };
-};
-
-&rcar_sound {
- ...
- ports {
- rsnd_port0: port@0 {
- rsnd_ep0: endpoint {
- remote-endpoint = <&codec_0>;
- ...
- playback = <&ssiu30 &ssi3>;
- };
- };
- rsnd_port1: port@1 {
- rsnd_ep1: endpoint {
- remote-endpoint = <&codec_1>;
- ...
- playback = <&ssiu31 &ssi3>;
- };
- };
- rsnd_port2: port@2 {
- rsnd_ep2: endpoint {
- remote-endpoint = <&codec_2>;
- ...
- playback = <&ssiu32 &ssi3>;
- };
- };
- rsnd_port3: port@3 {
- rsnd_ep3: endpoint {
- remote-endpoint = <&codec_3>;
- ...
- playback = <&ssiu33 &ssi3>;
- };
- };
- };
-};
-
-=============================================
-Example: simple sound card for Multi channel
-=============================================
-
-&rcar_sound {
- pinctrl-0 = <&sound_pins &sound_clk_pins>;
- pinctrl-names = "default";
-
- /* Single DAI */
- #sound-dai-cells = <0>;
-
-
- rcar_sound,dai {
- dai0 {
- playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>;
- };
- };
-};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
new file mode 100644
index 000000000000..0fd37aa84947
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -0,0 +1,447 @@
+# 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 Device Tree Bindings
+
+maintainers:
+ - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+
+ compatible:
+ oneOf:
+ # for Gen1 SoC
+ - items:
+ - enum:
+ - renesas,rcar_sound-r8a7778 # R-Car M1A
+ - renesas,rcar_sound-r8a7779 # R-Car H1
+ - enum:
+ - renesas,rcar_sound-gen1
+ # for Gen2 SoC
+ - items:
+ - enum:
+ - renesas,rcar_sound-r8a7742 # RZ/G1H
+ - renesas,rcar_sound-r8a7743 # RZ/G1M
+ - renesas,rcar_sound-r8a7744 # RZ/G1N
+ - renesas,rcar_sound-r8a7745 # RZ/G1E
+ - renesas,rcar_sound-r8a77470 # RZ/G1C
+ - renesas,rcar_sound-r8a7790 # R-Car H2
+ - renesas,rcar_sound-r8a7791 # R-Car M2-W
+ - renesas,rcar_sound-r8a7793 # R-Car M2-N
+ - renesas,rcar_sound-r8a7794 # R-Car E2
+ - enum:
+ - renesas,rcar_sound-gen2
+ # for Gen3 SoC
+ - items:
+ - enum:
+ - renesas,rcar_sound-r8a774a1 # RZ/G2M
+ - renesas,rcar_sound-r8a774b1 # RZ/G2N
+ - renesas,rcar_sound-r8a774c0 # RZ/G2E
+ - renesas,rcar_sound-r8a774e1 # RZ/G2H
+ - renesas,rcar_sound-r8a7795 # R-Car H3
+ - renesas,rcar_sound-r8a7796 # R-Car M3-W
+ - renesas,rcar_sound-r8a77961 # R-Car M3-W+
+ - renesas,rcar_sound-r8a77965 # R-Car M3-N
+ - renesas,rcar_sound-r8a77990 # R-Car E3
+ - renesas,rcar_sound-r8a77995 # R-Car D3
+ - enum:
+ - renesas,rcar_sound-gen3
+ # for Generic
+ - items:
+ - enum:
+ - renesas,rcar_sound-gen1
+ - renesas,rcar_sound-gen2
+ - renesas,rcar_sound-gen3
+
+ reg:
+ minItems: 1
+ maxItems: 5
+
+ reg-names:
+ minItems: 1
+ maxItems: 5
+
+ "#sound-dai-cells":
+ description: |
+ it must be 0 if your system is using single DAI
+ it must be 1 if your system is using multi DAIs
+ enum: [0, 1]
+
+ "#clock-cells":
+ description: |
+ it must be 0 if your system has audio_clkout
+ it must be 1 if your system has audio_clkout0/1/2/3
+ enum: [0, 1]
+
+ clock-frequency:
+ description: for audio_clkout0/1/2/3
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+
+ clkout-lr-asynchronous:
+ description: audio_clkoutn is asynchronizes with lr-clock.
+ $ref: /schemas/types.yaml#/definitions/flag
+
+ power-domains: true
+
+ resets:
+ maxItems: 11
+
+ reset-names:
+ maxItems: 11
+
+ clocks:
+ description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks.
+ minItems: 1
+ maxItems: 31
+
+ clock-names:
+ description: List of necessary clock names.
+ minItems: 1
+ maxItems: 31
+ items:
+ oneOf:
+ - const: ssi-all
+ - pattern: '^ssi\.[0-9]$'
+ - pattern: '^src\.[0-9]$'
+ - pattern: '^mix\.[0-1]$'
+ - pattern: '^ctu\.[0-1]$'
+ - pattern: '^dvc\.[0-1]$'
+ - pattern: '^clk_(a|b|c|i)$'
+
+ port: true
+
+# use patternProperties to avoid naming "xxx,yyy" issue
+patternProperties:
+ "^rcar_sound,dvc$":
+ description: DVC subnode.
+ type: object
+ patternProperties:
+ "^dvc-[0-1]$":
+ type: object
+ properties:
+ dmas:
+ maxItems: 1
+ dma-names:
+ const: "tx"
+ required:
+ - dmas
+ - dma-names
+ additionalProperties: false
+
+ "^rcar_sound,mix$":
+ description: MIX subnode.
+ type: object
+ patternProperties:
+ "^mix-[0-1]$":
+ type: object
+ # no properties
+ additionalProperties: false
+
+ "^rcar_sound,ctu$":
+ description: CTU subnode.
+ type: object
+ patternProperties:
+ "^ctu-[0-7]$":
+ type: object
+ # no properties
+ additionalProperties: false
+
+ "^rcar_sound,src$":
+ description: SRC subnode.
+ type: object
+ patternProperties:
+ "^src-[0-9]$":
+ type: object
+ properties:
+ interrupts:
+ maxItems: 1
+ dmas:
+ maxItems: 2
+ dma-names:
+ allOf:
+ - items:
+ enum:
+ - tx
+ - rx
+ required:
+ - interrupts
+ - dmas
+ - dma-names
+ additionalProperties: false
+
+ "^rcar_sound,ssiu$":
+ description: SSIU subnode.
+ type: object
+ patternProperties:
+ "^ssiu-[0-9]+$":
+ type: object
+ properties:
+ dmas:
+ maxItems: 2
+ dma-names:
+ allOf:
+ - items:
+ enum:
+ - tx
+ - rx
+ required:
+ - dmas
+ - dma-names
+ additionalProperties: false
+
+ "^rcar_sound,ssi$":
+ description: SSI subnode.
+ type: object
+ patternProperties:
+ "^ssi-[0-9]$":
+ type: object
+ properties:
+ interrupts:
+ maxItems: 1
+ dmas:
+ minItems: 2
+ maxItems: 4
+ dma-names:
+ allOf:
+ - items:
+ enum:
+ - tx
+ - rx
+ - txu # if no ssiu node
+ - rxu # if no ssiu node
+
+ shared-pin:
+ description: shared clock pin
+ $ref: /schemas/types.yaml#/definitions/flag
+ pio-transfer:
+ description: PIO transfer mode
+ $ref: /schemas/types.yaml#/definitions/flag
+ no-busif:
+ description: BUSIF is not used when [mem -> SSI] via DMA case
+ $ref: /schemas/types.yaml#/definitions/flag
+ required:
+ - interrupts
+ - dmas
+ - dma-names
+ additionalProperties: false
+
+ # For DAI base
+ "^rcar_sound,dai$":
+ description: DAI subnode.
+ type: object
+ patternProperties:
+ "^dai([0-9]+)?$":
+ type: object
+ properties:
+ playback:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ capture:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ anyOf:
+ - required:
+ - playback
+ - required:
+ - capture
+ additionalProperties: false
+
+required:
+ - compatible
+ - reg
+ - reg-names
+ - clocks
+ - clock-names
+ - "#sound-dai-cells"
+
+allOf:
+ - $ref: audio-graph.yaml#
+ - $ref: audio-graph-port.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: renesas,rcar_sound-gen1
+ then:
+ properties:
+ reg:
+ maxItems: 3
+ reg-names:
+ maxItems: 3
+ items:
+ enum:
+ - scu
+ - ssi
+ - adg
+ else:
+ properties:
+ reg:
+ maxItems: 5
+ reg-names:
+ maxItems: 5
+ items:
+ enum:
+ - scu
+ - adg
+ - ssiu
+ - ssi
+ - audmapp
+
+additionalProperties: false
+
+examples:
+ - |
+ rcar_sound: sound@ec500000 {
+ #sound-dai-cells = <1>;
+ compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2";
+ reg = <0xec500000 0x1000>, /* SCU */
+ <0xec5a0000 0x100>, /* ADG */
+ <0xec540000 0x1000>, /* SSIU */
+ <0xec541000 0x1280>, /* SSI */
+ <0xec740000 0x200>; /* Audio DMAC peri peri*/
+ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp";
+
+ clocks = <&mstp10_clks 1005>, /* SSI-ALL */
+ <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */
+ <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */
+ <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */
+ <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */
+ <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */
+ <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */
+ <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */
+ <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */
+ <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */
+ <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */
+ <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* MIX1, MIX0 */
+ <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CTU1, CTU0 */
+ <&mstp10_clks 1019>, <&mstp10_clks 1018>, /* DVC0, DVC1 */
+ <&audio_clk_a>, <&audio_clk_b>, /* CLKA, CLKB */
+ <&audio_clk_c>, <&audio_clk_i>; /* CLKC, CLKI */
+
+ clock-names = "ssi-all",
+ "ssi.9", "ssi.8",
+ "ssi.7", "ssi.6",
+ "ssi.5", "ssi.4",
+ "ssi.3", "ssi.2",
+ "ssi.1", "ssi.0",
+ "src.9", "src.8",
+ "src.7", "src.6",
+ "src.5", "src.4",
+ "src.3", "src.2",
+ "src.1", "src.0",
+ "mix.1", "mix.0",
+ "ctu.1", "ctu.0",
+ "dvc.0", "dvc.1",
+ "clk_a", "clk_b",
+ "clk_c", "clk_i";
+
+ rcar_sound,dvc {
+ dvc0: dvc-0 {
+ dmas = <&audma0 0xbc>;
+ dma-names = "tx";
+ };
+ dvc1: dvc-1 {
+ dmas = <&audma0 0xbe>;
+ dma-names = "tx";
+ };
+ };
+
+ rcar_sound,mix {
+ mix0: mix-0 { };
+ mix1: mix-1 { };
+ };
+
+ rcar_sound,ctu {
+ ctu00: ctu-0 { };
+ ctu01: ctu-1 { };
+ ctu02: ctu-2 { };
+ ctu03: ctu-3 { };
+ ctu10: ctu-4 { };
+ ctu11: ctu-5 { };
+ ctu12: ctu-6 { };
+ ctu13: ctu-7 { };
+ };
+
+ rcar_sound,src {
+ src0: src-0 {
+ status = "disabled";
+ };
+ src1: src-1 {
+ interrupts = <0 353 0>;
+ dmas = <&audma0 0x87>, <&audma1 0x9c>;
+ dma-names = "rx", "tx";
+ };
+ /* skip after src-2 */
+ };
+
+ rcar_sound,ssiu {
+ ssiu00: ssiu-0 {
+ dmas = <&audma0 0x15>, <&audma1 0x16>;
+ dma-names = "rx", "tx";
+ };
+ ssiu01: ssiu-1 {
+ dmas = <&audma0 0x35>, <&audma1 0x36>;
+ dma-names = "rx", "tx";
+ };
+ /* skip after ssiu-2 */
+ };
+
+ rcar_sound,ssi {
+ ssi0: ssi-0 {
+ interrupts = <0 370 1>;
+ dmas = <&audma0 0x01>, <&audma1 0x02>;
+ dma-names = "rx", "tx";
+ };
+ ssi1: ssi-1 {
+ interrupts = <0 371 1>;
+ dmas = <&audma0 0x03>, <&audma1 0x04>;
+ dma-names = "rx", "tx";
+ };
+ /* skip other ssi-2 */
+ };
+
+ /* DAI base */
+ rcar_sound,dai {
+ dai0 {
+ playback = <&ssi5 &src5>;
+ capture = <&ssi6>;
+ };
+ dai1 {
+ playback = <&ssi3>;
+ };
+ dai2 {
+ capture = <&ssi4>;
+ };
+ dai3 {
+ playback = <&ssi7>;
+ };
+ dai4 {
+ capture = <&ssi8>;
+ };
+ };
+
+ /* assume audio-graph */
+ port {
+ rsnd_endpoint: endpoint {
+ remote-endpoint = <&codec_endpoint>;
+
+ dai-format = "left_j";
+ bitclock-master = <&rsnd_endpoint0>;
+ frame-master = <&rsnd_endpoint0>;
+
+ playback = <&ssi0 &src0 &dvc0>;
+ capture = <&ssi1 &src1 &dvc1>;
+ };
+ };
+ };
+
+
+ /* assume audio-graph */
+ codec {
+ port {
+ codec_endpoint: endpoint {
+ remote-endpoint = <&rsnd_endpoint>;
+ };
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt
index 707fa98d1310..9c5fadb6ac82 100644
--- a/Documentation/devicetree/bindings/sound/rt5682.txt
+++ b/Documentation/devicetree/bindings/sound/rt5682.txt
@@ -44,6 +44,8 @@ Optional properties:
- realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of
the particular DMIC.
+- realtek,dmic-clk-driving-high : Set the high drving of the DMIC clock out.
+
Pins on the device (for linking into audio routes) for RT5682:
* DMIC L1
diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml
new file mode 100644
index 000000000000..5986d1fcbb54
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml
@@ -0,0 +1,41 @@
+# 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.
+
+properties:
+ compatible:
+ const: simple-audio-mux
+
+ mux-gpios:
+ description: |
+ GPIOs used to select the input line.
+
+ sound-name-prefix:
+ $ref: /schemas/types.yaml#/definitions/string
+ description:
+ Used as prefix for sink/source names of the component. Must be a
+ unique string among multiple instances of the same component.
+
+required:
+ - compatible
+ - mux-gpios
+
+additionalProperties: false
+
+examples:
+ - |
+ mux {
+ compatible = "simple-audio-mux";
+ mux-gpios = <&gpio 3 0>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml
index 35e669020296..45fd9fd9eb54 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.yaml
+++ b/Documentation/devicetree/bindings/sound/simple-card.yaml
@@ -13,13 +13,11 @@ definitions:
frame-master:
description: Indicates dai-link frame master.
- $ref: /schemas/types.yaml#/definitions/phandle-array
- maxItems: 1
+ $ref: /schemas/types.yaml#/definitions/phandle
bitclock-master:
description: Indicates dai-link bit clock master
- $ref: /schemas/types.yaml#/definitions/phandle-array
- maxItems: 1
+ $ref: /schemas/types.yaml#/definitions/phandle
frame-inversion:
description: dai-link uses frame clock inversion
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
deleted file mode 100644
index 864f5b00b031..000000000000
--- a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
+++ /dev/null
@@ -1,63 +0,0 @@
-STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM)
-
-The DFSDM allows PDM microphones capture through SPI interface. The Audio
-interface is seems as a sub block of the DFSDM device.
-For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.txt
-
-Required properties:
- - compatible: "st,stm32h7-dfsdm-dai".
-
- - #sound-dai-cells : Must be equal to 0
-
- - io-channels : phandle to iio dfsdm instance node.
-
-Example of a sound card using audio DFSDM node.
-
- sound_card {
- compatible = "audio-graph-card";
-
- dais = <&cpu_port>;
- };
-
- dfsdm: dfsdm@40017000 {
- compatible = "st,stm32h7-dfsdm";
- reg = <0x40017000 0x400>;
- clocks = <&rcc DFSDM1_CK>;
- clock-names = "dfsdm";
- #interrupt-cells = <1>;
- #address-cells = <1>;
- #size-cells = <0>;
-
- dfsdm_adc0: filter@0 {
- compatible = "st,stm32-dfsdm-dmic";
- reg = <0>;
- interrupts = <110>;
- dmas = <&dmamux1 101 0x400 0x00>;
- dma-names = "rx";
- st,adc-channels = <1>;
- st,adc-channel-names = "dmic0";
- st,adc-channel-types = "SPI_R";
- st,adc-channel-clk-src = "CLKOUT";
- st,filter-order = <5>;
-
- dfsdm_dai0: dfsdm-dai {
- compatible = "st,stm32h7-dfsdm-dai";
- #sound-dai-cells = <0>;
- io-channels = <&dfsdm_adc0 0>;
- cpu_port: port {
- dfsdm_endpoint: endpoint {
- remote-endpoint = <&dmic0_endpoint>;
- };
- };
- };
- };
-
- dmic0: dmic@0 {
- compatible = "dmic-codec";
- #sound-dai-cells = <0>;
- port {
- dmic0_endpoint: endpoint {
- remote-endpoint = <&dfsdm_endpoint>;
- };
- };
- };
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
deleted file mode 100644
index c42b91e525fa..000000000000
--- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
+++ /dev/null
@@ -1,107 +0,0 @@
-STMicroelectronics STM32 Serial Audio Interface (SAI).
-
-The SAI interface (Serial Audio Interface) offers a wide set of audio protocols
-as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97.
-The SAI contains two independent audio sub-blocks. Each sub-block has
-its own clock generator and I/O lines controller.
-
-Required properties:
- - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai"
- - reg: Base address and size of SAI common register set.
- - clocks: Must contain phandle and clock specifier pairs for each entry
- in clock-names.
- - clock-names: Must contain "pclk" "x8k" and "x11k"
- "pclk": Clock which feeds the peripheral bus interface.
- Mandatory for "st,stm32h7-sai" compatible.
- Not used for "st,stm32f4-sai" compatible.
- "x8k": SAI parent clock for sampling rates multiple of 8kHz.
- "x11k": SAI parent clock for sampling rates multiple of 11.025kHz.
- - interrupts: cpu DAI interrupt line shared by SAI sub-blocks
-
-Optional properties:
- - resets: Reference to a reset controller asserting the SAI
-
-SAI subnodes:
-Two subnodes corresponding to SAI sub-block instances A et B can be defined.
-Subnode can be omitted for unsused sub-block.
-
-SAI subnodes required properties:
- - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b"
- for SAI sub-block A or B respectively.
- - reg: Base address and size of SAI sub-block register set.
- - clocks: Must contain one phandle and clock specifier pair
- for sai_ck which feeds the internal clock generator.
- If the SAI shares a master clock, with another SAI set as MCLK
- clock provider, SAI provider phandle must be specified here.
- - clock-names: Must contain "sai_ck".
- Must also contain "MCLK", if SAI shares a master clock,
- with a SAI set as MCLK clock provider.
- - dmas: see Documentation/devicetree/bindings/dma/st,stm32-dma.yaml
- - dma-names: identifier string for each DMA request line
- "tx": if sai sub-block is configured as playback DAI
- "rx": if sai sub-block is configured as capture DAI
- - pinctrl-names: should contain only value "default"
- - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml
-
-SAI subnodes Optional properties:
- - st,sync: specify synchronization mode.
- By default SAI sub-block is in asynchronous mode.
- This property sets SAI sub-block as slave of another SAI sub-block.
- Must contain the phandle and index of the sai sub-block providing
- the synchronization.
- - st,iec60958: support S/PDIF IEC6958 protocol for playback
- IEC60958 protocol is not available for capture.
- By default, custom protocol is assumed, meaning that protocol is
- configured according to protocol defined in related DAI link node,
- such as i2s, left justified, right justified, dsp and pdm protocols.
- Note: ac97 protocol is not supported by SAI driver
- - #clock-cells: should be 0. This property must be present if the SAI device
- is a master clock provider, according to clocks bindings, described in
- Documentation/devicetree/bindings/clock/clock-bindings.txt.
-
-The device node should contain one 'port' child node with one child 'endpoint'
-node, according to the bindings defined in Documentation/devicetree/bindings/
-graph.txt.
-
-Example:
-sound_card {
- compatible = "audio-graph-card";
- dais = <&sai1b_port>;
-};
-
-sai1: sai1@40015800 {
- compatible = "st,stm32h7-sai";
- #address-cells = <1>;
- #size-cells = <1>;
- ranges = <0 0x40015800 0x400>;
- reg = <0x40015800 0x4>;
- clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>;
- clock-names = "pclk", "x8k", "x11k";
- interrupts = <87>;
-
- sai1a: audio-controller@40015804 {
- compatible = "st,stm32-sai-sub-a";
- reg = <0x4 0x1C>;
- clocks = <&rcc SAI1_CK>;
- clock-names = "sai_ck";
- dmas = <&dmamux1 1 87 0x400 0x0>;
- dma-names = "tx";
- pinctrl-names = "default";
- pinctrl-0 = <&pinctrl_sai1a>;
-
- sai1b_port: port {
- cpu_endpoint: endpoint {
- remote-endpoint = <&codec_endpoint>;
- format = "i2s";
- };
- };
- };
-};
-
-audio-codec {
- codec_port: port {
- codec_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint>;
- };
- };
-};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
new file mode 100644
index 000000000000..6ad48c7681c1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.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/st,stm32-sai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: STMicroelectronics STM32 Serial Audio Interface (SAI)
+
+maintainers:
+ - Olivier Moysan <olivier.moysan@st.com>
+
+description:
+ The SAI interface (Serial Audio Interface) offers a wide set of audio
+ protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97.
+ The SAI contains two independent audio sub-blocks. Each sub-block has
+ its own clock generator and I/O lines controller.
+
+properties:
+ compatible:
+ enum:
+ - st,stm32f4-sai
+ - st,stm32h7-sai
+
+ reg:
+ items:
+ - description: Base address and size of SAI common register set.
+ - description: Base address and size of SAI identification register set.
+ minItems: 1
+ maxItems: 2
+
+ ranges:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 1
+
+ clocks:
+ maxItems: 3
+
+ clock-names:
+ maxItems: 3
+
+required:
+ - compatible
+ - reg
+ - ranges
+ - "#address-cells"
+ - "#size-cells"
+ - clocks
+ - clock-names
+
+patternProperties:
+ "^audio-controller@[0-9a-f]+$":
+ type: object
+ description:
+ Two subnodes corresponding to SAI sub-block instances A et B
+ can be defined. Subnode can be omitted for unsused sub-block.
+
+ properties:
+ compatible:
+ description: Compatible for SAI sub-block A or B.
+ pattern: "st,stm32-sai-sub-[ab]"
+
+ "#sound-dai-cells":
+ const: 0
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: sai_ck clock feeding the internal clock generator.
+ - description: MCLK clock from a SAI set as master clock provider.
+ minItems: 1
+ maxItems: 2
+
+ clock-names:
+ items:
+ - const: sai_ck
+ - const: MCLK
+ minItems: 1
+ maxItems: 2
+
+ 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.
+ allOf:
+ - $ref: /schemas/types.yaml#definitions/phandle-array
+ - maxItems: 1
+
+ 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.
+ allOf:
+ - $ref: /schemas/types.yaml#definitions/flag
+
+ "#clock-cells":
+ description: Configure the SAI device as master clock provider.
+ const: 0
+
+ required:
+ - compatible
+ - "#sound-dai-cells"
+ - reg
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+allOf:
+ - if:
+ properties:
+ compatible:
+ contains:
+ const: st,stm32f4-sai
+
+ - then:
+ properties:
+ clocks:
+ items:
+ - description: x8k, SAI parent clock for sampling rates multiple of 8kHz.
+ - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz.
+
+ clock-names:
+ items:
+ - const: x8k
+ - const: x11k
+
+ - else:
+ properties:
+ clocks:
+ items:
+ - description: pclk feeds the peripheral bus interface.
+ - description: x8k, SAI parent clock for sampling rates multiple of 8kHz.
+ - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz.
+
+ clock-names:
+ items:
+ - const: pclk
+ - const: x8k
+ - const: x11k
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/stm32mp1-clks.h>
+ #include <dt-bindings/reset/stm32mp1-resets.h>
+ sai2: sai@4400b000 {
+ compatible = "st,stm32h7-sai";
+ #address-cells = <1>;
+ #size-cells = <1>;
+ ranges = <0 0x4400b000 0x400>;
+ reg = <0x4400b000 0x4>, <0x4400b3f0 0x10>;
+ clocks = <&rcc SAI2>, <&rcc PLL3_Q>, <&rcc PLL3_R>;
+ clock-names = "pclk", "x8k", "x11k";
+ pinctrl-names = "default", "sleep";
+ pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>;
+ pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>;
+ status = "okay";
+
+ 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";
+ status = "okay";
+ };
+ };
+
+...
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst
index c755b1c5e16f..fe52c314b763 100644
--- a/Documentation/sound/alsa-configuration.rst
+++ b/Documentation/sound/alsa-configuration.rst
@@ -2227,6 +2227,11 @@ quirk_alias
Quirk alias list, pass strings like ``0123abcd:5678beef``, which
applies the existing quirk for the device 5678:beef to a new
device 0123:abcd.
+implicit_fb
+ Apply the generic implicit feedback sync mode. When this is set
+ and the playback stream sync mode is ASYNC, the driver tries to
+ tie an adjacent ASYNC capture stream as the implicit feedback
+ source.
use_vmalloc
Use vmalloc() for allocations of the PCM buffers (default: yes).
For architectures with non-coherent memory like ARM or MIPS, the