Merge tag 'sound-6.12-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "A fairly big update at this time, both in core and driver sides.

  The core received rewrites in PCM buffer allocation handling and
  locking optimizations, PCM rate updates followed by lots of cleanups.

  In ASoC side, the legacy Intel drivers have been deprecated by AVS
  drivers which leaded to the significant amount of code reduction.
  SoundWire driver updates and other cleanups contributed more code
  reduction, too.

  USB-audio driver received a large cleanup of its big quirk table, and
  the old snd_print*() API usages in many legacy drivers are replaced
  with the standard print API.

  Here are some highlights:

  Core:
   - More optimized locking in ALSA control code
   - Rewrites of memalloc helpers for better DMA API usage
   - Drop of obsoleted vmalloc PCM buffer helper API
   - Continued MIDI2 UMP updates
   - Support of a new user-space driven timer instance
   - Update for more PCM support rates and cleanups
   - Xrun counter report in the proc files

  ASoC:
   - Continued simplification and cleanup works for ASoC
   - Extensive cleanups and refactoring of the Soundwire drivers
   - 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

  USB-audio:
   - Add support of multiple control interfaces
   - A large rewrite of quirk table with macros
   - Support for RME Digiface USB

  HD-audio:
   - Cleanup of quirk code for Samsung Galaxy laptops
   - Clean up of detection of Cirrus codecs
   - C-Media CM9825 HD-audio codec support

  Others:
   - Rewrites to standard print API in a lot of legacy drivers"

* tag 'sound-6.12-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (410 commits)
  ASoC: topology: Fix redundant logical jump
  ASoC: tas2781: Add Calibration Kcontrols for Chromebook
  ASoC: amd: acp: refactor SoundWire machine driver code
  ASoC: sdw_utils/intel: move soundwire endpoint parsing helper functions
  ASoC: sdw_util/intel: move soundwire endpoint and dai link structures
  ASoC: intel: sof_sdw: rename soundwire parsing helper functions
  ASoC: intel: sof_sdw: rename soundwire endpoint and dailink structures
  ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls
  ALSA: hda/realtek: Add support for Galaxy Book2 Pro (NP950XEE)
  ASoC: mediatek: mt7986-afe-pcm: Remove redundant error message
  ALSA: memalloc: Use proper DMA mapping API for x86 S/G buffer allocations
  ALSA: memalloc: Use proper DMA mapping API for x86 WC buffer allocations
  ALSA: usb-audio: Add logitech Audio profile quirk
  ASoc: mediatek: mt8365: Remove unneeded assignment
  ASoC: Intel: ARL: Add entry for HDMI-In capture support to non-I2S codec boards.
  ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for ARL.
  ASoC: SOF: Intel: hda: remove common_hdmi_codec_drv
  ASoC: Intel: sof_pcm512x: do not check common_hdmi_codec_drv
  ASoC: Intel: ehl_rt5660: do not check common_hdmi_codec_drv
  ASoC: Intel: skl_hda_dsp_generic: use common module for DAI links
  ...

32 files changed, 1316 insertions, 554 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>;
-};
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst
index 829c672d9fe6..04254474fa04 100644
--- a/Documentation/sound/alsa-configuration.rst
+++ b/Documentation/sound/alsa-configuration.rst
@@ -1059,6 +1059,9 @@ power_save
     Automatic power-saving timeout (in second, 0 = disable)
 power_save_controller
     Reset HD-audio controller in power-saving mode (default = on)
+pm_blacklist
+    Enable / disable power-management deny-list (default = look up PM
+    deny-list, 0 = skip PM deny-list, 1 = force to turn off runtime PM)
 align_buffer_size
     Force rounding of buffer/period sizes to multiples of 128 bytes.
     This is more efficient in terms of memory access but isn't
diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst
index ef6a4513cce7..e199131bf5ab 100644
--- a/Documentation/sound/hd-audio/notes.rst
+++ b/Documentation/sound/hd-audio/notes.rst
@@ -321,12 +321,6 @@ Kernel Configuration
 --------------------
 In general, I recommend you to enable the sound debug option,
 ``CONFIG_SND_DEBUG=y``, no matter whether you are debugging or not.
-This enables snd_printd() macro and others, and you'll get additional
-kernel messages at probing.
-
-In addition, you can enable ``CONFIG_SND_DEBUG_VERBOSE=y``.  But this
-will give you far more messages.  Thus turn this on only when you are
-sure to want it.
 
 Don't forget to turn on the appropriate ``CONFIG_SND_HDA_CODEC_*``
 options.  Note that each of them corresponds to the codec chip, not
diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst
index 7e67e12730d3..c437f2a4bc85 100644
--- a/Documentation/sound/index.rst
+++ b/Documentation/sound/index.rst
@@ -13,6 +13,7 @@ Sound Subsystem Documentation
    alsa-configuration
    hd-audio/index
    cards/index
+   utimers
 
 .. only::  subproject and html
 
diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
index 801b0bb57e97..895752cbcedd 100644
--- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
+++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
@@ -4030,31 +4030,6 @@ located in the new subdirectory, sound/pci/xyz.
 Useful Functions
 ================
 
-:c:func:`snd_printk()` and friends
-----------------------------------
-
-.. note:: This subsection describes a few helper functions for
-   decorating a bit more on the standard :c:func:`printk()` & co.
-   However, in general, the use of such helpers is no longer recommended.
-   If possible, try to stick with the standard functions like
-   :c:func:`dev_err()` or :c:func:`pr_err()`.
-
-ALSA provides a verbose version of the :c:func:`printk()` function.
-If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function
-prints the given message together with the file name and the line of the
-caller. The ``KERN_XXX`` prefix is processed as well as the original
-:c:func:`printk()` does, so it's recommended to add this prefix,
-e.g. snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\\n");
-
-There are also :c:func:`printk()`'s for debugging.
-:c:func:`snd_printd()` can be used for general debugging purposes.
-If ``CONFIG_SND_DEBUG`` is set, this function is compiled, and works
-just like :c:func:`snd_printk()`. If the ALSA is compiled without
-the debugging flag, it's ignored.
-
-:c:func:`snd_printdd()` is compiled in only when
-``CONFIG_SND_DEBUG_VERBOSE`` is set.
-
 :c:func:`snd_BUG()`
 -------------------
 
diff --git a/Documentation/sound/utimers.rst b/Documentation/sound/utimers.rst
new file mode 100644
index 000000000000..ec21567d3f72
--- /dev/null
+++ b/Documentation/sound/utimers.rst
@@ -0,0 +1,126 @@
+.. SPDX-License-Identifier: GPL-2.0
+
+=======================
+Userspace-driven timers
+=======================
+
+:Author: Ivan Orlov <ivan.orlov0322@gmail.com>
+
+Preface
+=======
+
+This document describes the userspace-driven timers: virtual ALSA timers
+which could be created and controlled by userspace applications using
+IOCTL calls. Such timers could be useful when synchronizing audio
+stream with timer sources which we don't have ALSA timers exported for
+(e.g. PTP clocks), and when synchronizing the audio stream going through
+two virtual sound devices using ``snd-aloop`` (for instance, when
+we have a network application sending frames to one snd-aloop device,
+and another sound application listening on the other end of snd-aloop).
+
+Enabling userspace-driven timers
+================================
+
+The userspace-driven timers could be enabled in the kernel using the
+``CONFIG_SND_UTIMER`` configuration option. It depends on the
+``CONFIG_SND_TIMER`` option, so it also should be enabled.
+
+Userspace-driven timers API
+===========================
+
+Userspace application can create a userspace-driven ALSA timer by
+executing the ``SNDRV_TIMER_IOCTL_CREATE`` ioctl call on the
+``/dev/snd/timer`` device file descriptor. The ``snd_timer_uinfo``
+structure should be passed as an ioctl argument:
+
+::
+
+    struct snd_timer_uinfo {
+        __u64 resolution;
+        int fd;
+        unsigned int id;
+        unsigned char reserved[16];
+    }
+
+The ``resolution`` field sets the desired resolution in nanoseconds for
+the virtual timer. ``resolution`` field simply provides an information
+about the virtual timer, but does not affect the timing itself. ``id``
+field gets overwritten by the ioctl, and the identifier you get in this
+field after the call can be used as a timer subdevice number when
+passing the timer to ``snd-aloop`` kernel module or other userspace
+applications. There could be up to 128 userspace-driven timers in the
+system at one moment of time, thus the id value ranges from 0 to 127.
+
+Besides from overwriting the ``snd_timer_uinfo`` struct, ioctl stores
+a timer file descriptor, which can be used to trigger the timer, in the
+``fd`` field of the ``snd_timer_uinfo`` struct. Allocation of a file
+descriptor for the timer guarantees that the timer can only be triggered
+by the process which created it. The timer then can be triggered with
+``SNDRV_TIMER_IOCTL_TRIGGER`` ioctl call on the timer file descriptor.
+
+So, the example code for creating and triggering the timer would be:
+
+::
+
+    static struct snd_timer_uinfo utimer_info = {
+        /* Timer is going to tick (presumably) every 1000000 ns */
+        .resolution = 1000000ULL,
+        .id = -1,
+    };
+
+    int timer_device_fd = open("/dev/snd/timer",  O_RDWR | O_CLOEXEC);
+
+    if (ioctl(timer_device_fd, SNDRV_TIMER_IOCTL_CREATE, &utimer_info)) {
+        perror("Failed to create the timer");
+        return -1;
+    }
+
+    ...
+
+    /*
+     * Now we want to trigger the timer. Callbacks of all of the
+     * timer instances binded to this timer will be executed after
+     * this call.
+     */
+    ioctl(utimer_info.fd, SNDRV_TIMER_IOCTL_TRIGGER, NULL);
+
+    ...
+
+    /* Now, destroy the timer */
+    close(timer_info.fd);
+
+
+More detailed example of creating and ticking the timer could be found
+in the utimer ALSA selftest.
+
+Userspace-driven timers and snd-aloop
+-------------------------------------
+
+Userspace-driven timers could be easily used with ``snd-aloop`` module
+when synchronizing two sound applications on both ends of the virtual
+sound loopback. For instance, if one of the applications receives sound
+frames from network and sends them to snd-aloop pcm device, and another
+application listens for frames on the other snd-aloop pcm device, it
+makes sense that the ALSA middle layer should initiate a data
+transaction when the new period of data is received through network, but
+not when the certain amount of jiffies elapses. Userspace-driven ALSA
+timers could be used to achieve this.
+
+To use userspace-driven ALSA timer as a timer source of snd-aloop, pass
+the following string as the snd-aloop ``timer_source`` parameter:
+
+::
+
+  # modprobe snd-aloop timer_source="-1.4.<utimer_id>"
+
+Where ``utimer_id`` is the id of the timer you created with
+``SNDRV_TIMER_IOCTL_CREATE``, and ``4`` is the number of
+userspace-driven timers device (``SNDRV_TIMER_GLOBAL_UDRIVEN``).
+
+``resolution`` for the userspace-driven ALSA timer used with snd-aloop
+should be calculated as ``1000000000ULL / frame_rate * period_size`` as
+the timer is going to tick every time a new period of frames is ready.
+
+After that, each time you trigger the timer with
+``SNDRV_TIMER_IOCTL_TRIGGER`` the new period of data will be transferred
+from one snd-aloop device to another.