Merge tag 'sound-5.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been some significant changes in the core side, both for
  ALSA and ASoC, while lots of development have been seen in SOF, as
  well as many small fixes/improvements for ASoC codecs and platforms.
  Below is a highlight in this cycle:

  Core:
   - The unification of PCM vmalloc buffer allocation helpers into the
     standard API
   - Clean up of the default PCM mmap handling for vmalloc & SG-buffer
   - Fix potential races at ALSA timer open
   - A few new PCM API extensions; just preliminary core changes, the
     actual changes in drivers will be merged in 5.6
   - Continued ASoC componentization works; now almost everything is a
     common ASoC component object. A lot of refactoring and
     simplification have been done along with it.

  ASoC:
   - Many fixes to the Sound Open Firmware (SOF) code
   - Wake on voice support for Chromebooks
   - SPI support and trigger word detection for RT5677
   - New drivers for Analog Devices ADAU7118, Intel Cannonlake systems
     with RT1011 and RT5682, Texas Instruments TAS2562 and TAS2770

  HD-audio:
   - Improved Intel DSP configuration / probe code for SOF
   - Plumbing the legacy HD-audio driver with Intel SOF HDMI
   - DP-MST support for Nvidia HDMI codecs
   - Realtek quirks cleanups and new additions as usual

  Others:
   - Lots of refactoring and cleanups for FireWire; period-size sharing,
     h/w IRQ interval configuration, clock recovery improvements, etc
   - USB-audio: Scarlett mixer quirks
   - Cleanups of PCM calls in various drivers (including media and USB)
     to adapt the core API changes"

* tag 'sound-5.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (497 commits)
  ALSA: usb-audio: Fix Focusrite Scarlett 6i6 gen1 - input handling
  ALSA: hda/realtek - Enable internal speaker of ASUS UX431FLC
  ALSA: aloop: Fix dependency on timer API
  ASoC: DMI long name - avoid to add board name if matches with product name
  ASoC: improve the DMI long card code in asoc-core
  ASoC: rsnd: fix DALIGN register for SSIU
  ALSA: aloop: Avoid unexpected timer event callback tasklets
  ALSA: aloop: Remove redundant locking in timer open function
  ASoC: component: Add sync_stop PCM ops
  ASoC: pcm: Make ioctl ops optional
  ALSA: hda/hdmi - Clear codec->relaxed_resume flag at unbinding
  ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs
  ALSA: cs4236: fix error return comparison of an unsigned integer
  ALSA: usb-audio: Fix NULL dereference at parsing BADD
  ALSA: usb-audio: Fix Scarlett 6i6 Gen 2 port data
  ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop
  ALSA: hda/realtek - Move some alc236 pintbls to fallback table
  ALSA: hda/realtek - Move some alc256 pintbls to fallback table
  ALSA: docs: Update about the new PCM sync_stop ops
  ALSA: pcm: Add card sync_irq field
  ...

26 files changed, 1045 insertions, 360 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml
new file mode 100644
index 000000000000..75e0cbe6be70
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml
@@ -0,0 +1,85 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau7118.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+
+title: Analog Devices ADAU7118 8 Channel PDM to I2S/TDM Converter
+
+maintainers:
+  - Nuno Sá <nuno.sa@analog.com>
+
+description: |
+  Analog Devices ADAU7118 8 Channel PDM to I2S/TDM Converter over I2C or HW
+  standalone mode.
+  https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU7118.pdf
+
+properties:
+  compatible:
+    enum:
+      - adi,adau7118
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  iovdd-supply:
+    description: Digital Input/Output Power Supply.
+
+  dvdd-supply:
+    description: Internal Core Digital Power Supply.
+
+  adi,decimation-ratio:
+    description: |
+      This property set's the decimation ratio of PDM to PCM audio data.
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+      - enum: [64, 32, 16]
+        default: 64
+
+  adi,pdm-clk-map:
+    description: |
+      The ADAU7118 has two PDM clocks for the four Inputs. Each input must be
+      assigned to one of these two clocks. This property set's the mapping
+      between the clocks and the inputs.
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32-array
+      - minItems: 4
+        maxItems: 4
+        items:
+          maximum: 1
+        default: [0, 0, 1, 1]
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - iovdd-supply
+  - dvdd-supply
+
+examples:
+  - |
+    i2c {
+        /* example with i2c support */
+        #address-cells = <1>;
+        #size-cells = <0>;
+        adau7118_codec: audio-codec@14 {
+                compatible = "adi,adau7118";
+                reg = <0x14>;
+                #sound-dai-cells = <0>;
+                iovdd-supply = <&supply>;
+                dvdd-supply = <&supply>;
+                adi,pdm-clk-map = <1 1 0 0>;
+                adi,decimation-ratio = <16>;
+        };
+    };
+
+    /* example with hw standalone mode */
+    adau7118_codec_hw: adau7118-codec-hw {
+            compatible = "adi,adau7118";
+            #sound-dai-cells = <0>;
+            iovdd-supply = <&supply>;
+            dvdd-supply = <&supply>;
+    };
diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml
new file mode 100644
index 000000000000..b8f89c7258eb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml
@@ -0,0 +1,267 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner A10 Codec Device Tree Bindings
+
+maintainers:
+  - Chen-Yu Tsai <wens@csie.org>
+  - Maxime Ripard <maxime.ripard@bootlin.com>
+
+properties:
+  "#sound-dai-cells":
+    const: 0
+
+  compatible:
+    enum:
+      - allwinner,sun4i-a10-codec
+      - allwinner,sun6i-a31-codec
+      - allwinner,sun7i-a20-codec
+      - allwinner,sun8i-a23-codec
+      - allwinner,sun8i-h3-codec
+      - allwinner,sun8i-v3s-codec
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Bus Clock
+      - description: Module Clock
+
+  clock-names:
+    items:
+      - const: apb
+      - const: codec
+
+  dmas:
+    items:
+      - description: RX DMA Channel
+      - description: TX DMA Channel
+
+  dma-names:
+    items:
+      - const: rx
+      - const: tx
+
+  resets:
+    maxItems: 1
+
+  allwinner,audio-routing:
+    description: |-
+      A list of the connections between audio components.  Each entry
+      is a pair of strings, the first being the connection's sink, the
+      second being the connection's source.
+    allOf:
+      - $ref: /schemas/types.yaml#definitions/non-unique-string-array
+      - minItems: 2
+        maxItems: 18
+        items:
+          enum:
+            # Audio Pins on the SoC
+            - HP
+            - HPCOM
+            - LINEIN
+            - LINEOUT
+            - MIC1
+            - MIC2
+            - MIC3
+
+            # Microphone Biases from the SoC
+            - HBIAS
+            - MBIAS
+
+            # Board Connectors
+            - Headphone
+            - Headset Mic
+            - Line In
+            - Line Out
+            - Mic
+            - Speaker
+
+  allwinner,codec-analog-controls:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: Phandle to the codec analog controls in the PRCM
+
+  allwinner,pa-gpios:
+    description: GPIO to enable the external amplifier
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+
+allOf:
+  - if:
+      properties:
+        compatible:
+          enum:
+            - allwinner,sun6i-a31-codec
+            - allwinner,sun8i-a23-codec
+            - allwinner,sun8i-h3-codec
+            - allwinner,sun8i-v3s-codec
+
+    then:
+      if:
+        properties:
+          compatible:
+            const: allwinner,sun6i-a31-codec
+
+      then:
+        required:
+          - resets
+          - allwinner,audio-routing
+
+      else:
+        required:
+          - resets
+          - allwinner,audio-routing
+          - allwinner,codec-analog-controls
+
+  - if:
+      properties:
+        compatible:
+          enum:
+            - allwinner,sun6i-a31-codec
+
+    then:
+      properties:
+        allwinner,audio-routing:
+          items:
+            enum:
+              - HP
+              - HPCOM
+              - LINEIN
+              - LINEOUT
+              - MIC1
+              - MIC2
+              - MIC3
+              - HBIAS
+              - MBIAS
+              - Headphone
+              - Headset Mic
+              - Line In
+              - Line Out
+              - Mic
+              - Speaker
+
+  - if:
+      properties:
+        compatible:
+          enum:
+            - allwinner,sun8i-a23-codec
+
+    then:
+      properties:
+        allwinner,audio-routing:
+          items:
+            enum:
+              - HP
+              - HPCOM
+              - LINEIN
+              - MIC1
+              - MIC2
+              - HBIAS
+              - MBIAS
+              - Headphone
+              - Headset Mic
+              - Line In
+              - Line Out
+              - Mic
+              - Speaker
+
+  - if:
+      properties:
+        compatible:
+          enum:
+            - allwinner,sun8i-h3-codec
+
+    then:
+      properties:
+        allwinner,audio-routing:
+          items:
+            enum:
+              - HP
+              - HPCOM
+              - LINEIN
+              - LINEOUT
+              - MIC1
+              - MIC2
+              - HBIAS
+              - MBIAS
+              - Headphone
+              - Headset Mic
+              - Line In
+              - Line Out
+              - Mic
+              - Speaker
+
+  - if:
+      properties:
+        compatible:
+          enum:
+            - allwinner,sun8i-v3s-codec
+
+    then:
+      properties:
+        allwinner,audio-routing:
+          items:
+            enum:
+              - HP
+              - HPCOM
+              - MIC1
+              - HBIAS
+              - Headphone
+              - Headset Mic
+              - Line In
+              - Line Out
+              - Mic
+              - Speaker
+
+additionalProperties: false
+
+examples:
+  - |
+    codec@1c22c00 {
+        #sound-dai-cells = <0>;
+        compatible = "allwinner,sun7i-a20-codec";
+        reg = <0x01c22c00 0x40>;
+        interrupts = <0 30 4>;
+        clocks = <&apb0_gates 0>, <&codec_clk>;
+        clock-names = "apb", "codec";
+        dmas = <&dma 0 19>, <&dma 0 19>;
+        dma-names = "rx", "tx";
+    };
+
+  - |
+    codec@1c22c00 {
+        #sound-dai-cells = <0>;
+        compatible = "allwinner,sun6i-a31-codec";
+        reg = <0x01c22c00 0x98>;
+        interrupts = <0 29 4>;
+        clocks = <&ccu 61>, <&ccu 135>;
+        clock-names = "apb", "codec";
+        resets = <&ccu 42>;
+        dmas = <&dma 15>, <&dma 15>;
+        dma-names = "rx", "tx";
+        allwinner,audio-routing =
+            "Headphone", "HP",
+            "Speaker", "LINEOUT",
+            "LINEIN", "Line In",
+            "MIC1", "MBIAS",
+            "MIC1", "Mic",
+            "MIC2", "HBIAS",
+            "MIC2", "Headset Mic";
+    };
+
+...
diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml
new file mode 100644
index 000000000000..85305b4c2729
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a23-codec-analog.yaml
@@ -0,0 +1,38 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a23-codec-analog.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner A23 Analog Codec Device Tree Bindings
+
+maintainers:
+  - Chen-Yu Tsai <wens@csie.org>
+  - Maxime Ripard <maxime.ripard@bootlin.com>
+
+properties:
+  compatible:
+    enum:
+      # FIXME: This is documented in the PRCM binding, but needs to be
+      # migrated here at some point
+      # - allwinner,sun8i-a23-codec-analog
+      - allwinner,sun8i-h3-codec-analog
+      - allwinner,sun8i-v3s-codec-analog
+
+  reg:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    codec_analog: codec-analog@1f015c0 {
+      compatible = "allwinner,sun8i-h3-codec-analog";
+      reg = <0x01f015c0 0x4>;
+    };
+
+...
diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt
index 0e76946385ae..17530120ccfc 100644
--- a/Documentation/devicetree/bindings/sound/arndale.txt
+++ b/Documentation/devicetree/bindings/sound/arndale.txt
@@ -1,8 +1,9 @@
 Audio Binding for Arndale boards
 
 Required properties:
-- compatible : Can be the following,
-			"samsung,arndale-rt5631"
+- compatible : Can be one of the following:
+		"samsung,arndale-rt5631",
+		"samsung,arndale-wm1811"
 
 - samsung,audio-cpu: The phandle of the Samsung I2S controller
 - samsung,audio-codec: The phandle of the audio codec
diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt
new file mode 100644
index 000000000000..40353fc30255
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt
@@ -0,0 +1,36 @@
+fsl,mqs audio CODEC
+
+Required properties:
+  - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs"
+		"fsl,imx8qm-mqs", "fsl,imx8qxp-mqs".
+  - clocks : A list of phandles + clock-specifiers, one for each entry in
+	     clock-names
+  - clock-names : "mclk" - must required.
+		  "core" - required if compatible is "fsl,imx8qm-mqs", it
+		           is for register access.
+  - gpr : A phandle of General Purpose Registers in IOMUX Controller.
+	  Required if compatible is "fsl,imx6sx-mqs".
+
+Required if compatible is "fsl,imx8qm-mqs":
+  - power-domains: A phandle of PM domain provider node.
+  - reg: Offset and length of the register set for the device.
+
+Example:
+
+mqs: mqs {
+	compatible = "fsl,imx6sx-mqs";
+	gpr = <&gpr>;
+	clocks = <&clks IMX6SX_CLK_SAI1>;
+	clock-names = "mclk";
+	status = "disabled";
+};
+
+mqs: mqs@59850000 {
+	compatible = "fsl,imx8qm-mqs";
+	reg = <0x59850000 0x10000>;
+	clocks = <&clk IMX8QM_AUD_MQS_IPG>,
+		 <&clk IMX8QM_AUD_MQS_HMCLK>;
+	clock-names = "core", "mclk";
+	power-domains = <&pd_mqs0>;
+	status = "disabled";
+};
diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt
index 1084f7f22eea..8ca52dcc5572 100644
--- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt
+++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt
@@ -1,4 +1,4 @@
-* Audio codec controlled by ChromeOS EC
+Audio codec controlled by ChromeOS EC
 
 Google's ChromeOS EC codec is a digital mic codec provided by the
 Embedded Controller (EC) and is controlled via a host-command interface.
@@ -9,10 +9,27 @@ Documentation/devicetree/bindings/mfd/cros-ec.txt).
 Required properties:
 - compatible: Must contain "google,cros-ec-codec"
 - #sound-dai-cells: Should be 1. The cell specifies number of DAIs.
-- max-dmic-gain: A number for maximum gain in dB on digital microphone.
+
+Optional properties:
+- reg: Pysical base address and length of shared memory region from EC.
+       It contains 3 unsigned 32-bit integer.  The first 2 integers
+       combine to become an unsigned 64-bit physical address.  The last
+       one integer is length of the shared memory.
+- memory-region: Shared memory region to EC.  A "shared-dma-pool".  See
+                 ../reserved-memory/reserved-memory.txt for details.
 
 Example:
 
+{
+	...
+
+	reserved_mem: reserved_mem {
+		compatible = "shared-dma-pool";
+		reg = <0 0x52800000 0 0x100000>;
+		no-map;
+	};
+}
+
 cros-ec@0 {
 	compatible = "google,cros-ec-spi";
 
@@ -21,6 +38,7 @@ cros-ec@0 {
 	cros_ec_codec: ec-codec {
 		compatible = "google,cros-ec-codec";
 		#sound-dai-cells = <1>;
-		max-dmic-gain = <43>;
+		reg = <0x0 0x10500000 0x80000>;
+		memory-region = <&reserved_mem>;
 	};
 };
diff --git a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt
index 396ba38619f6..1f1cba4152ce 100644
--- a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt
+++ b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt
@@ -4,6 +4,10 @@ Required properties:
 - compatible = "mediatek,mt68183-audio";
 - reg: register location and size
 - interrupts: should contain AFE interrupt
+- resets: Must contain an entry for each entry in reset-names
+  See ../reset/reset.txt for details.
+- reset-names: should have these reset names:
+		"audiosys";
 - power-domains: should define the power domain
 - clocks: Must contain an entry for each entry in clock-names
 - clock-names: should have these clock names:
@@ -20,6 +24,8 @@ Example:
 		compatible = "mediatek,mt8183-audio";
 		reg = <0 0x11220000 0 0x1000>;
 		interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>;
+		resets = <&watchdog MT8183_TOPRGU_AUDIO_SW_RST>;
+		reset-names = "audiosys";
 		power-domains = <&scpsys MT8183_POWER_DOMAIN_AUDIO>;
 		clocks = <&infrasys CLK_INFRA_AUDIO>,
 			 <&infrasys CLK_INFRA_AUDIO_26M_BCLK>,
diff --git a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt
index d6d5207fa996..decaa013a07e 100644
--- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt
+++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt
@@ -2,14 +2,19 @@ MT8183 with MT6358, TS3A227 and MAX98357 CODECS
 
 Required properties:
 - compatible : "mediatek,mt8183_mt6358_ts3a227_max98357"
-- mediatek,headset-codec: the phandles of ts3a227 codecs
 - mediatek,platform: the phandle of MT8183 ASoC platform
 
+Optional properties:
+- mediatek,headset-codec: the phandles of ts3a227 codecs
+- mediatek,ec-codec: the phandle of EC codecs.
+                     See google,cros-ec-codec.txt for more details.
+
 Example:
 
 	sound {
 		compatible = "mediatek,mt8183_mt6358_ts3a227_max98357";
 		mediatek,headset-codec = <&ts3a227>;
+		mediatek,ec-codec = <&ec_codec>;
 		mediatek,platform = <&afe>;
 	};
 
diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
deleted file mode 100644
index 0cf0f819b823..000000000000
--- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt
+++ /dev/null
@@ -1,31 +0,0 @@
-Renesas FSI
-
-Required properties:
-- compatible			: "renesas,fsi2-<soctype>",
-				  "renesas,sh_fsi2" or "renesas,sh_fsi" as
-				  fallback.
-				  Examples with soctypes are:
-				    - "renesas,fsi2-r8a7740" (R-Mobile A1)
-				    - "renesas,fsi2-sh73a0" (SH-Mobile AG5)
-- reg				: Should contain the register physical address and length
-- interrupts			: Should contain FSI interrupt
-
-- fsia,spdif-connection		: FSI is connected by S/PDIF
-- fsia,stream-mode-support	: FSI supports 16bit stream mode.
-- fsia,use-internal-clock	: FSI uses internal clock when master mode.
-
-- fsib,spdif-connection		: same as fsia
-- fsib,stream-mode-support	: same as fsia
-- fsib,use-internal-clock	: same as fsia
-
-Example:
-
-sh_fsi2: sh_fsi2@ec230000 {
-	compatible = "renesas,sh_fsi2";
-	reg = <0xec230000 0x400>;
-	interrupts = <0 146 0x4>;
-
-	fsia,spdif-connection;
-	fsia,stream-mode-support;
-	fsia,use-internal-clock;
-};
diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml
new file mode 100644
index 000000000000..140a37fc3c0b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml
@@ -0,0 +1,76 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/renesas,fsi.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Renesas FSI Sound Driver Device Tree Bindings
+
+maintainers:
+  - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+  $nodename:
+    pattern: "^sound@.*"
+
+  compatible:
+    oneOf:
+      # for FSI2 SoC
+      - items:
+        - enum:
+          - renesas,fsi2-sh73a0
+          - renesas,fsi2-r8a7740
+        - enum:
+          - renesas,sh_fsi2
+      # for Generic
+      - items:
+        - enum:
+          - renesas,sh_fsi
+          - renesas,sh_fsi2
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  fsia,spdif-connection:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description: FSI is connected by S/PDIF
+
+  fsia,stream-mode-support:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description: FSI supports 16bit stream mode
+
+  fsia,use-internal-clock:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description: FSI uses internal clock when master mode
+
+  fsib,spdif-connection:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description: same as fsia
+
+  fsib,stream-mode-support:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description: same as fsia
+
+  fsib,use-internal-clock:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description: same as fsia
+
+required:
+  - compatible
+  - reg
+  - interrupts
+
+examples:
+  - |
+    sh_fsi2: sound@ec230000 {
+            compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2";
+            reg = <0xec230000 0x400>;
+            interrupts = <0 146 0x4>;
+
+            fsia,spdif-connection;
+            fsia,stream-mode-support;
+            fsia,use-internal-clock;
+    };
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 5c52182f7dcf..797fd035434c 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -268,6 +268,7 @@ Required properties:
 				    - "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-r8a7778" (R-Car M1A)
 				    - "renesas,rcar_sound-r8a7779" (R-Car H1)
diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98090.txt b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt
index a805aa99ad75..e9c58b204399 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-max98090.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt
@@ -5,15 +5,38 @@ Required properties:
 - rockchip,model: The user-visible name of this sound complex
 - rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's
   connected to the CODEC
-- rockchip,audio-codec: The phandle of the MAX98090 audio codec
-- rockchip,headset-codec: The phandle of Ext chip for jack detection
+
+Optional properties:
+- rockchip,audio-codec: The phandle of the MAX98090 audio codec.
+- rockchip,headset-codec: The phandle of Ext chip for jack detection. This is
+                          required if there is rockchip,audio-codec.
+- rockchip,hdmi-codec: The phandle of HDMI device for HDMI codec.
 
 Example:
 
+/* For max98090-only board. */
+sound {
+	compatible = "rockchip,rockchip-audio-max98090";
+	rockchip,model = "ROCKCHIP-I2S";
+	rockchip,i2s-controller = <&i2s>;
+	rockchip,audio-codec = <&max98090>;
+	rockchip,headset-codec = <&headsetcodec>;
+};
+
+/* For HDMI-only board. */
+sound {
+	compatible = "rockchip,rockchip-audio-max98090";
+	rockchip,model = "ROCKCHIP-I2S";
+	rockchip,i2s-controller = <&i2s>;
+	rockchip,hdmi-codec = <&hdmi>;
+};
+
+/* For max98090 plus HDMI board. */
 sound {
 	compatible = "rockchip,rockchip-audio-max98090";
 	rockchip,model = "ROCKCHIP-I2S";
 	rockchip,i2s-controller = <&i2s>;
 	rockchip,audio-codec = <&max98090>;
 	rockchip,headset-codec = <&headsetcodec>;
+	rockchip,hdmi-codec = <&hdmi>;
 };
diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt
index 35a23e60d679..02d53b9aa247 100644
--- a/Documentation/devicetree/bindings/sound/rt1011.txt
+++ b/Documentation/devicetree/bindings/sound/rt1011.txt
@@ -20,6 +20,14 @@ Required properties:
         |     1     |    1     |   0x3b     |
         -------------------------------------
 
+Optional properties:
+
+- realtek,temperature_calib
+  u32. The temperature was measured while doing the calibration. Units: Celsius degree
+
+- realtek,r0_calib
+  u32. This is r0 calibration data which was measured in factory mode.
+
 Pins on the device (for linking into audio routes) for RT1011:
 
   * SPO
@@ -29,4 +37,6 @@ Example:
 rt1011: codec@38 {
 	compatible = "realtek,rt1011";
 	reg = <0x38>;
+	realtek,temperature_calib = <25>;
+	realtek,r0_calib = <0x224050>;
 };
diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt
index 312e9a129530..30e927a28369 100644
--- a/Documentation/devicetree/bindings/sound/rt5682.txt
+++ b/Documentation/devicetree/bindings/sound/rt5682.txt
@@ -27,6 +27,11 @@ Optional properties:
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
+- realtek,btndet-delay
+  The debounce delay for push button.
+  The delay time is realtek,btndet-delay value multiple of 8.192 ms.
+  If absent, the default is 16.
+
 Pins on the device (for linking into audio routes) for RT5682:
 
   * DMIC L1
@@ -47,4 +52,5 @@ rt5682 {
 	realtek,dmic1-data-pin = <1>;
 	realtek,dmic1-clk-pin = <1>;
 	realtek,jd-src = <1>;
+	realtek,btndet-delay = <16>;
 };
diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt
deleted file mode 100644
index e9da2200e173..000000000000
--- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt
+++ /dev/null
@@ -1,54 +0,0 @@
-Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec
-
-Required properties:
-
- - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board,
-		"hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated),
-		"samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated),
-		"samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated)
- - model - the user-visible name of this sound complex
- - clocks - should contain entries matching clock names in the clock-names
-    property
- - samsung,audio-widgets - this property specifies off-codec audio elements
-   like headphones or speakers, for details see widgets.txt
- - samsung,audio-routing - a list of the connections between audio
-   components;  each entry is a pair of strings, the first being the
-   connection's sink, the second being the connection's source;
-   valid names for sources and sinks are the MAX98090's pins (as
-   documented in its binding), and the jacks on the board
-
-   For Odroid X2:
-     "Headphone Jack", "Mic Jack", "DMIC"
-
-   For Odroid U3, XU3:
-     "Headphone Jack", "Speakers"
-
-   For Odroid XU4:
-     no entries
-
-Required sub-nodes:
-
- - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
-    controller
- - 'codec' subnode with a 'sound-dai' property containing list of phandles
-    to the CODEC nodes, first entry must be corresponding to the MAX98090
-    CODEC and the second entry must be the phandle of the HDMI IP block node
-
-Example:
-
-sound {
-	compatible = "hardkernel,odroid-xu3-audio";
-	model = "Odroid-XU3";
-	samsung,audio-routing =
-		"Headphone Jack", "HPL",
-		"Headphone Jack", "HPR",
-		"IN1", "Mic Jack",
-		"Mic Jack", "MICBIAS";
-
-	cpu {
-		sound-dai = <&i2s0 0>;
-	};
-	codec {
-		sound-dai = <&hdmi>, <&max98090>;
-	};
-};
diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml
new file mode 100644
index 000000000000..c6b244352d05
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml
@@ -0,0 +1,91 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec
+
+maintainers:
+  - Krzysztof Kozlowski <krzk@kernel.org>
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    oneOf:
+      - const: hardkernel,odroid-xu3-audio
+
+      - const: hardkernel,odroid-xu4-audio
+        deprecated: true
+
+      - const: samsung,odroid-xu3-audio
+        deprecated: true
+
+      - const: samsung,odroid-xu4-audio
+        deprecated: true
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: The user-visible name of this sound complex.
+
+  cpu:
+    type: object
+    properties:
+      sound-dai:
+        $ref: /schemas/types.yaml#/definitions/phandle-array
+        description: phandles to the I2S controllers
+
+  codec:
+    type: object
+    properties:
+      sound-dai:
+        $ref: /schemas/types.yaml#/definitions/phandle-array
+        description: |
+          List of phandles to the CODEC nodes,
+          first entry must be corresponding to the MAX98090 CODEC and
+          the second entry must be the phandle of the HDMI IP block node.
+
+  samsung,audio-routing:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description: |
+      List of the connections between audio
+      components;  each entry is a pair of strings, the first being the
+      connection's sink, the second being the connection's source;
+      valid names for sources and sinks are the MAX98090's pins (as
+      documented in its binding), and the jacks on the board.
+      For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC"
+      For Odroid U3, XU3: "Headphone Jack", "Speakers"
+      For Odroid XU4: no entries
+
+  samsung,audio-widgets:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description: |
+      This property specifies off-codec audio elements
+      like headphones or speakers, for details see widgets.txt
+
+required:
+  - compatible
+  - model
+  - cpu
+  - codec
+
+examples:
+  - |
+    sound {
+        compatible = "hardkernel,odroid-xu3-audio";
+        model = "Odroid-XU3";
+        samsung,audio-routing =
+                "Headphone Jack", "HPL",
+                "Headphone Jack", "HPR",
+                "IN1", "Mic Jack",
+                "Mic Jack", "MICBIAS";
+
+        cpu {
+            sound-dai = <&i2s0 0>;
+        };
+
+        codec {
+            sound-dai = <&hdmi>, <&max98090>;
+        };
+    };
+
diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
deleted file mode 100644
index a88cb00fa096..000000000000
--- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt
+++ /dev/null
@@ -1,84 +0,0 @@
-* Samsung I2S controller
-
-Required SoC Specific Properties:
-
-- compatible : should be one of the following.
-   - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
-   - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with
-     secondary fifo, s/w reset control and internal mux for root clk src.
-   - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for
-     playback, stereo channel capture, secondary fifo using internal
-     or external dma, s/w reset control, internal mux for root clk src
-     and 7.1 channel TDM support for playback. TDM (Time division multiplexing)
-     is to allow transfer of multiple channel audio data on single data line.
-   - samsung,exynos7-i2s: with all the available features of exynos5 i2s,
-     exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo
-     with only external dma and more no.of root clk sampling frequencies.
-   - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports
-     stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with
-     slightly modified bit offsets.
-
-- reg: physical base address of the controller and length of memory mapped
-  region.
-- dmas: list of DMA controller phandle and DMA request line ordered pairs.
-- dma-names: identifier string for each DMA request line in the dmas property.
-  These strings correspond 1:1 with the ordered pairs in dmas.
-- clocks: Handle to iis clock and RCLK source clk.
-- clock-names:
-  i2s0 uses some base clocks from CMU and some are from audio subsystem internal
-  clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and
-  "i2s_opclk1" as shown in the example below.
-  i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should
-  be "iis" and "i2s_opclk0".
-  "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root
-  clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2
-  doesn't have any such mux.
-- #clock-cells: should be 1, this property must be present if the I2S device
-  is a clock provider in terms of the common clock bindings, described in
-  ../clock/clock-bindings.txt.
-- clock-output-names (deprecated): from the common clock bindings, names of
-  the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1",
-  "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively.
-
-There are following clocks available at the I2S device nodes:
- CLK_I2S_CDCLK    - the CDCLK (CODECLKO) gate clock,
- CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the
-		    IISPSR register),
- CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in
-		    IISMOD register).
-
-Refer to the SoC datasheet for availability of the above clocks.
-The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available
-in the IIS Multi Audio Interface.
-
-Note: Old DTs may not have the #clock-cells property and then not use the I2S
-node as a clock supplier.
-
-Optional SoC Specific Properties:
-
-- samsung,idma-addr: Internal DMA register base address of the audio
-  sub system(used in secondary sound source).
-- pinctrl-0: Should specify pin control groups used for this controller.
-- pinctrl-names: Should contain only one value - "default".
-- #sound-dai-cells: should be 1.
-
-
-Example:
-
-i2s0: i2s@3830000 {
-	compatible = "samsung,s5pv210-i2s";
-	reg = <0x03830000 0x100>;
-	dmas = <&pdma0 10
-		&pdma0 9
-		&pdma0 8>;
-	dma-names = "tx", "rx", "tx-sec";
-	clocks = <&clock_audss EXYNOS_I2S_BUS>,
-		<&clock_audss EXYNOS_I2S_BUS>,
-		<&clock_audss EXYNOS_SCLK_I2S>;
-	clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
-	#clock-cells = <1>;
-	samsung,idma-addr = <0x03000000>;
-	pinctrl-names = "default";
-	pinctrl-0 = <&i2s0_bus>;
-	#sound-dai-cells = <1>;
-};
diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml
new file mode 100644
index 000000000000..53e3bad4178c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml
@@ -0,0 +1,138 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung SoC I2S controller
+
+maintainers:
+  - Krzysztof Kozlowski <krzk@kernel.org>
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    description: |
+      samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
+
+      samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with
+      secondary FIFO, s/w reset control and internal mux for root clock
+      source.
+
+      samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for
+      playback, stereo channel capture, secondary FIFO using internal
+      or external DMA, s/w reset control, internal mux for root clock
+      source and 7.1 channel TDM support for playback; TDM (Time division
+      multiplexing) is to allow transfer of multiple channel audio data on
+      single data line.
+
+      samsung,exynos7-i2s: with all the available features of Exynos5 I2S.
+      Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO
+      with only external DMA and more number of root clock sampling
+      frequencies.
+
+      samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports
+      stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with
+      slightly modified bit offsets.
+    enum:
+      - samsung,s3c6410-i2s
+      - samsung,s5pv210-i2s
+      - samsung,exynos5420-i2s
+      - samsung,exynos7-i2s
+      - samsung,exynos7-i2s1
+
+  reg:
+    maxItems: 1
+
+  dmas:
+    minItems: 2
+    maxItems: 3
+
+  dma-names:
+    oneOf:
+      - items:
+          - const: tx
+          - const: rx
+      - items:
+          - const: tx
+          - const: rx
+          - const: tx-sec
+
+  clocks:
+    minItems: 1
+    maxItems: 3
+
+  clock-names:
+    oneOf:
+      - items:
+          - const: iis
+      - items: # for I2S0
+          - const: iis
+          - const: i2s_opclk0
+          - const: i2s_opclk1
+      - items: # for I2S1 and I2S2
+          - const: iis
+          - const: i2s_opclk0
+    description: |
+      "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources
+      of the root clock. I2S0 has internal mux to select the source
+      of root clock and I2S1 and I2S2 doesn't have any such mux.
+
+  "#clock-cells":
+    const: 1
+
+  clock-output-names:
+    deprecated: true
+    oneOf:
+      - items: # for I2S0
+          - const: i2s_cdclk0
+      - items: # for I2S1
+          - const: i2s_cdclk1
+      - items: # for I2S2
+          - const: i2s_cdclk2
+    description: Names of the CDCLK I2S output clocks.
+
+  samsung,idma-addr:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: |
+      Internal DMA register base address of the audio
+      subsystem (used in secondary sound source).
+
+  pinctrl-0:
+    description: Should specify pin control groups used for this controller.
+
+  pinctrl-names:
+    const: default
+
+  "#sound-dai-cells":
+    const: 1
+
+required:
+  - compatible
+  - reg
+  - dmas
+  - dma-names
+  - clocks
+  - clock-names
+
+examples:
+  - |
+    #include <dt-bindings/clock/exynos-audss-clk.h>
+
+    i2s0: i2s@3830000 {
+        compatible = "samsung,s5pv210-i2s";
+        reg = <0x03830000 0x100>;
+        dmas = <&pdma0 10>,
+                <&pdma0 9>,
+                <&pdma0 8>;
+        dma-names = "tx", "rx", "tx-sec";
+        clocks = <&clock_audss EXYNOS_I2S_BUS>,
+                <&clock_audss EXYNOS_I2S_BUS>,
+                <&clock_audss EXYNOS_SCLK_I2S>;
+        clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
+        #clock-cells = <1>;
+        samsung,idma-addr = <0x03000000>;
+        pinctrl-names = "default";
+        pinctrl-0 = <&i2s0_bus>;
+        #sound-dai-cells = <1>;
+    };
diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
deleted file mode 100644
index 66579bbd3294..000000000000
--- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt
+++ /dev/null
@@ -1,94 +0,0 @@
-* Allwinner A10 Codec
-
-Required properties:
-- compatible: must be one of the following compatibles:
-		- "allwinner,sun4i-a10-codec"
-		- "allwinner,sun6i-a31-codec"
-		- "allwinner,sun7i-a20-codec"
-		- "allwinner,sun8i-a23-codec"
-		- "allwinner,sun8i-h3-codec"
-		- "allwinner,sun8i-v3s-codec"
-- reg: must contain the registers location and length
-- interrupts: must contain the codec interrupt
-- dmas: DMA channels for tx and rx dma. See the DMA client binding,
-	Documentation/devicetree/bindings/dma/dma.txt
-- dma-names: should include "tx" and "rx".
-- clocks: a list of phandle + clock-specifer pairs, one for each entry
-  in clock-names.
-- clock-names: should contain the following:
-   - "apb": the parent APB clock for this controller
-   - "codec": the parent module clock
-
-Optional properties:
-- allwinner,pa-gpios: gpio to enable external amplifier
-
-Required properties for the following compatibles:
-		- "allwinner,sun6i-a31-codec"
-		- "allwinner,sun8i-a23-codec"
-		- "allwinner,sun8i-h3-codec"
-		- "allwinner,sun8i-v3s-codec"
-- resets: phandle to the reset control for this device
-- allwinner,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 include:
-
-			   Audio pins on the SoC:
-			   "HP"
-			   "HPCOM"
-			   "LINEIN"	(not on sun8i-v3s)
-			   "LINEOUT"	(not on sun8i-a23 or sun8i-v3s)
-			   "MIC1"
-			   "MIC2"	(not on sun8i-v3s)
-			   "MIC3"	(sun6i-a31 only)
-
-			   Microphone biases from the SoC:
-			   "HBIAS"
-			   "MBIAS"	(not on sun8i-v3s)
-
-			   Board connectors:
-			   "Headphone"
-			   "Headset Mic"
-			   "Line In"
-			   "Line Out"
-			   "Mic"
-			   "Speaker"
-
-Required properties for the following compatibles:
-		- "allwinner,sun8i-a23-codec"
-		- "allwinner,sun8i-h3-codec"
-		- "allwinner,sun8i-v3s-codec"
-- allwinner,codec-analog-controls: A phandle to the codec analog controls
-				   block in the PRCM.
-
-Example:
-codec: codec@1c22c00 {
-	#sound-dai-cells = <0>;
-	compatible = "allwinner,sun7i-a20-codec";
-	reg = <0x01c22c00 0x40>;
-	interrupts = <0 30 4>;
-	clocks = <&apb0_gates 0>, <&codec_clk>;
-	clock-names = "apb", "codec";
-	dmas = <&dma 0 19>, <&dma 0 19>;
-	dma-names = "rx", "tx";
-};
-
-codec: codec@1c22c00 {
-	#sound-dai-cells = <0>;
-	compatible = "allwinner,sun6i-a31-codec";
-	reg = <0x01c22c00 0x98>;
-	interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>;
-	clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>;
-	clock-names = "apb", "codec";
-	resets = <&ccu RST_APB1_CODEC>;
-	dmas = <&dma 15>, <&dma 15>;
-	dma-names = "rx", "tx";
-	allwinner,audio-routing =
-		"Headphone", "HP",
-		"Speaker", "LINEOUT",
-		"LINEIN", "Line In",
-		"MIC1",	"MBIAS",
-		"MIC1", "Mic",
-		"MIC2", "HBIAS",
-		"MIC2", "Headset Mic";
-};
diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt
deleted file mode 100644
index 07356758bd91..000000000000
--- a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt
+++ /dev/null
@@ -1,17 +0,0 @@
-* Allwinner Codec Analog Controls
-
-Required properties:
-- compatible: must be one of the following compatibles:
-		- "allwinner,sun8i-a23-codec-analog"
-		- "allwinner,sun8i-h3-codec-analog"
-		- "allwinner,sun8i-v3s-codec-analog"
-
-Required properties if not a sub-node of the PRCM node:
-- reg: must contain the registers location and length
-
-Example:
-prcm: prcm@1f01400 {
-	codec_analog: codec-analog {
-		compatible = "allwinner,sun8i-a23-codec-analog";
-	};
-};
diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt
new file mode 100644
index 000000000000..658e1fb18a99
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas2562.txt
@@ -0,0 +1,34 @@
+Texas Instruments TAS2562 Smart PA
+
+The TAS2562 is a mono, digital input Class-D audio amplifier optimized for
+efficiently driving high peak power into small loudspeakers.
+Integrated speaker voltage and current sense provides for
+real time monitoring of loudspeaker behavior.
+
+Required properties:
+ - #address-cells  - Should be <1>.
+ - #size-cells     - Should be <0>.
+ - compatible:	   - Should contain "ti,tas2562".
+ - reg:		   - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f.
+ - ti,imon-slot-no:- TDM TX current sense time slot.
+
+Optional properties:
+- interrupt-parent: phandle to the interrupt controller which provides
+                    the interrupt.
+- interrupts: (GPIO) interrupt to which the chip is connected.
+- shut-down: GPIO used to control the state of the device.
+
+Examples:
+tas2562@4c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        compatible = "ti,tas2562";
+        reg = <0x4c>;
+
+        interrupt-parent = <&gpio1>;
+        interrupts = <14>;
+
+	shut-down = <&gpio1 15 0>;
+        ti,imon-slot-no = <0>;
+};
+
diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt
new file mode 100644
index 000000000000..ede6bb3d9637
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas2770.txt
@@ -0,0 +1,37 @@
+Texas Instruments TAS2770 Smart PA
+
+The TAS2770 is a mono, digital input Class-D audio amplifier optimized for
+efficiently driving high peak power into small loudspeakers.
+Integrated speaker voltage and current sense provides for
+real time monitoring of loudspeaker behavior.
+
+Required properties:
+
+ - compatible:	   - Should contain "ti,tas2770".
+ - reg:		       - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>.
+ - #address-cells  - Should be <1>.
+ - #size-cells     - Should be <0>.
+ - ti,asi-format:  - Sets TDM RX capture edge. 0->Rising; 1->Falling.
+ - ti,imon-slot-no:- TDM TX current sense time slot.
+ - ti,vmon-slot-no:- TDM TX voltage sense time slot.
+
+Optional properties:
+
+- interrupt-parent: the phandle to the interrupt controller which provides
+                     the interrupt.
+- interrupts: interrupt specification for data-ready.
+
+Examples:
+
+    tas2770@4c {
+                compatible = "ti,tas2770";
+                reg = <0x4c>;
+                #address-cells = <1>;
+                #size-cells = <0>;
+                interrupt-parent = <&msm_gpio>;
+                interrupts = <97 0>;
+                ti,asi-format = <0>;
+                ti,imon-slot-no = <0>;
+                ti,vmon-slot-no = <2>;
+        };
+
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt
index 5d9cb84c661d..a02ecaab5183 100644
--- a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt
+++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt
@@ -25,6 +25,13 @@ Required properties:
 
 For required properties on SPI/I2C, consult SPI/I2C device tree documentation
 
+Optional properties:
+
+  - reset-gpios : Optional reset gpio line connected to RST pin of the codec.
+		  The RST line is low active:
+		  RST = low: device power-down
+		  RST = high: device is enabled
+
 Examples:
 
 i2c0: i2c0@0 {
@@ -34,6 +41,7 @@ i2c0: i2c0@0 {
 	pcm3168a: audio-codec@44 {
 		compatible = "ti,pcm3168a";
 		reg = <0x44>;
+		reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>;
 		clocks = <&clk_core CLK_AUDIO>;
 		clock-names = "scki";
 		VDD1-supply = <&supply3v3>;
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
index 5b3c33bb99e5..e372303697dc 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@@ -29,6 +29,11 @@ Optional properties:
         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:
 
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.yaml b/Documentation/devicetree/bindings/vendor-prefixes.yaml
index 05b3904a995b..fd6fa07c45b8 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.yaml
+++ b/Documentation/devicetree/bindings/vendor-prefixes.yaml
@@ -16,7 +16,7 @@ properties: {}
 patternProperties:
   # Prefixes which are not vendors, but followed the pattern
   # DO NOT ADD NEW PROPERTIES TO THIS LIST
-  "^(at25|devbus|dmacap|dsa|exynos|gpio-fan|gpio|gpmc|hdmi|i2c-gpio),.*": true
+  "^(at25|devbus|dmacap|dsa|exynos|fsi[ab]|gpio-fan|gpio|gpmc|hdmi|i2c-gpio),.*": true
   "^(keypad|m25p|max8952|max8997|max8998|mpmc),.*": true
   "^(pinctrl-single|#pinctrl-single|PowerPC),.*": true
   "^(pl022|pxa-mmc|rcar_sound|rotary-encoder|s5m8767|sdhci),.*": true
diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
index 132f5eb9b530..f169d58ca019 100644
--- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
+++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
@@ -805,6 +805,7 @@ destructor and PCI entries. Example code is shown first, below.
                       return -EBUSY;
               }
               chip->irq = pci->irq;
+              card->sync_irq = chip->irq;
 
               /* (2) initialization of the chip hardware */
               .... /*   (not implemented in this document) */
@@ -965,6 +966,15 @@ usually like the following:
           return IRQ_HANDLED;
   }
 
+After requesting the IRQ, you can passed it to ``card->sync_irq``
+field:
+::
+
+          card->irq = chip->irq;
+
+This allows PCM core automatically performing
+:c:func:`synchronize_irq()` at the necessary timing like ``hw_free``.
+See the later section `sync_stop callback`_ for details.
 
 Now let's write the corresponding destructor for the resources above.
 The role of destructor is simple: disable the hardware (if already
@@ -1270,21 +1280,23 @@ shows only the skeleton, how to build up the PCM interfaces.
               /* the hardware-specific codes will be here */
               ....
               return 0;
-
       }
 
       /* hw_params callback */
       static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream,
                                    struct snd_pcm_hw_params *hw_params)
       {
-              return snd_pcm_lib_malloc_pages(substream,
-                                         params_buffer_bytes(hw_params));
+              /* the hardware-specific codes will be here */
+              ....
+              return 0;
       }
 
       /* hw_free callback */
       static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream)
       {
-              return snd_pcm_lib_free_pages(substream);
+              /* the hardware-specific codes will be here */
+              ....
+              return 0;
       }
 
       /* prepare callback */
@@ -1339,7 +1351,6 @@ shows only the skeleton, how to build up the PCM interfaces.
       static struct snd_pcm_ops snd_mychip_playback_ops = {
               .open =        snd_mychip_playback_open,
               .close =       snd_mychip_playback_close,
-              .ioctl =       snd_pcm_lib_ioctl,
               .hw_params =   snd_mychip_pcm_hw_params,
               .hw_free =     snd_mychip_pcm_hw_free,
               .prepare =     snd_mychip_pcm_prepare,
@@ -1351,7 +1362,6 @@ shows only the skeleton, how to build up the PCM interfaces.
       static struct snd_pcm_ops snd_mychip_capture_ops = {
               .open =        snd_mychip_capture_open,
               .close =       snd_mychip_capture_close,
-              .ioctl =       snd_pcm_lib_ioctl,
               .hw_params =   snd_mychip_pcm_hw_params,
               .hw_free =     snd_mychip_pcm_hw_free,
               .prepare =     snd_mychip_pcm_prepare,
@@ -1382,9 +1392,9 @@ shows only the skeleton, how to build up the PCM interfaces.
                               &snd_mychip_capture_ops);
               /* pre-allocation of buffers */
               /* NOTE: this may fail */
-              snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
-                                                    snd_dma_pci_data(chip->pci),
-                                                    64*1024, 64*1024);
+              snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
+                                             &chip->pci->dev,
+                                             64*1024, 64*1024);
               return 0;
       }
 
@@ -1454,7 +1464,6 @@ The operators are defined typically like this:
   static struct snd_pcm_ops snd_mychip_playback_ops = {
           .open =        snd_mychip_pcm_open,
           .close =       snd_mychip_pcm_close,
-          .ioctl =       snd_pcm_lib_ioctl,
           .hw_params =   snd_mychip_pcm_hw_params,
           .hw_free =     snd_mychip_pcm_hw_free,
           .prepare =     snd_mychip_pcm_prepare,
@@ -1465,13 +1474,14 @@ The operators are defined typically like this:
 All the callbacks are described in the Operators_ subsection.
 
 After setting the operators, you probably will want to pre-allocate the
-buffer. For the pre-allocation, simply call the following:
+buffer and set up the managed allocation mode.
+For that, simply call the following:
 
 ::
 
-  snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
-                                        snd_dma_pci_data(chip->pci),
-                                        64*1024, 64*1024);
+  snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
+                                 &chip->pci->dev,
+                                 64*1024, 64*1024);
 
 It will allocate a buffer up to 64kB as default. Buffer management
 details will be described in the later section `Buffer and Memory
@@ -1621,8 +1631,7 @@ For the operators (callbacks) of each sound driver, most of these
 records are supposed to be read-only. Only the PCM middle-layer changes
 / updates them. The exceptions are the hardware description (hw) DMA
 buffer information and the private data. Besides, if you use the
-standard buffer allocation method via
-:c:func:`snd_pcm_lib_malloc_pages()`, you don't need to set the
+standard managed buffer allocation mode, you don't need to set the
 DMA buffer information by yourself.
 
 In the sections below, important records are explained.
@@ -1776,8 +1785,8 @@ the physical address of the buffer. This field is specified only when
 the buffer is a linear buffer. ``dma_bytes`` holds the size of buffer
 in bytes. ``dma_private`` is used for the ALSA DMA allocator.
 
-If you use a standard ALSA function,
-:c:func:`snd_pcm_lib_malloc_pages()`, for allocating the buffer,
+If you use either the managed buffer allocation mode or the standard
+API function :c:func:`snd_pcm_lib_malloc_pages()` for allocating the buffer,
 these fields are set by the ALSA middle layer, and you should *not*
 change them by yourself. You can read them but not write them. On the
 other hand, if you want to allocate the buffer by yourself, you'll
@@ -1911,7 +1920,10 @@ ioctl callback
 ~~~~~~~~~~~~~~
 
 This is used for any special call to pcm ioctls. But usually you can
-pass a generic ioctl callback, :c:func:`snd_pcm_lib_ioctl()`.
+leave it as NULL, then PCM core calls the generic ioctl callback
+function :c:func:`snd_pcm_lib_ioctl()`.  If you need to deal with the
+unique setup of channel info or reset procedure, you can pass your own
+callback function here.
 
 hw_params callback
 ~~~~~~~~~~~~~~~~~~~
@@ -1929,8 +1941,12 @@ Many hardware setups should be done in this callback, including the
 allocation of buffers.
 
 Parameters to be initialized are retrieved by
-:c:func:`params_xxx()` macros. To allocate buffer, you can call a
-helper function,
+:c:func:`params_xxx()` macros.
+
+When you set up the managed buffer allocation mode for the substream,
+a buffer is already allocated before this callback gets
+called. Alternatively, you can call a helper function below for
+allocating the buffer, too.
 
 ::
 
@@ -1964,18 +1980,23 @@ hw_free callback
   static int snd_xxx_hw_free(struct snd_pcm_substream *substream);
 
 This is called to release the resources allocated via
-``hw_params``. For example, releasing the buffer via
-:c:func:`snd_pcm_lib_malloc_pages()` is done by calling the
-following:
-
-::
-
-  snd_pcm_lib_free_pages(substream);
+``hw_params``.
 
 This function is always called before the close callback is called.
 Also, the callback may be called multiple times, too. Keep track
 whether the resource was already released.
 
+When you have set up the managed buffer allocation mode for the PCM
+substream, the allocated PCM buffer will be automatically released
+after this callback gets called.  Otherwise you'll have to release the
+buffer manually.  Typically, when the buffer was allocated from the
+pre-allocated pool, you can use the standard API function
+:c:func:`snd_pcm_lib_malloc_pages()` like:
+
+::
+
+  snd_pcm_lib_free_pages(substream);
+
 prepare callback
 ~~~~~~~~~~~~~~~~
 
@@ -2048,6 +2069,37 @@ flag set, and you cannot call functions which may sleep. The
 triggering the DMA. The other stuff should be initialized
 ``hw_params`` and ``prepare`` callbacks properly beforehand.
 
+sync_stop callback
+~~~~~~~~~~~~~~~~~~
+
+::
+
+  static int snd_xxx_sync_stop(struct snd_pcm_substream *substream);
+
+This callback is optional, and NULL can be passed.  It's called after
+the PCM core stops the stream and changes the stream state
+``prepare``, ``hw_params`` or ``hw_free``.
+Since the IRQ handler might be still pending, we need to wait until
+the pending task finishes before moving to the next step; otherwise it
+might lead to a crash due to resource conflicts or access to the freed
+resources.  A typical behavior is to call a synchronization function
+like :c:func:`synchronize_irq()` here.
+
+For majority of drivers that need only a call of
+:c:func:`synchronize_irq()`, there is a simpler setup, too.
+While keeping NULL to ``sync_stop`` PCM callback, the driver can set
+``card->sync_irq`` field to store the valid interrupt number after
+requesting an IRQ, instead.   Then PCM core will look call
+:c:func:`synchronize_irq()` with the given IRQ appropriately.
+
+If the IRQ handler is released at the card destructor, you don't need
+to clear ``card->sync_irq``, as the card itself is being released.
+So, usually you'll need to add just a single line for assigning
+``card->sync_irq`` in the driver code unless the driver re-acquires
+the IRQ.  When the driver frees and re-acquires the IRQ dynamically
+(e.g. for suspend/resume), it needs to clear and re-set
+``card->sync_irq`` again appropriately.
+
 pointer callback
 ~~~~~~~~~~~~~~~~
 
@@ -2095,10 +2147,12 @@ This callback is atomic as default.
 page callback
 ~~~~~~~~~~~~~
 
-This callback is optional too. This callback is used mainly for
-non-contiguous buffers. The mmap calls this callback to get the page
-address. Some examples will be explained in the later section `Buffer
-and Memory Management`_, too.
+This callback is optional too. The mmap calls this callback to get the
+page fault address.
+
+Since the recent changes, you need no special callback any longer for
+the standard SG-buffer or vmalloc-buffer. Hence this callback should
+be rarely used.
 
 mmap calllback
 ~~~~~~~~~~~~~~
@@ -3512,7 +3566,7 @@ bus).
 ::
 
   snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
-                                        snd_dma_pci_data(pci), size, max);
+                                        &pci->dev, size, max);
 
 where ``size`` is the byte size to be pre-allocated and the ``max`` is
 the maximum size to be changed via the ``prealloc`` proc file. The
@@ -3523,12 +3577,14 @@ The second argument (type) and the third argument (device pointer) are
 dependent on the bus. For normal devices, pass the device pointer
 (typically identical as ``card->dev``) to the third argument with
 ``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the
-bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type and the
-``snd_dma_continuous_data(GFP_KERNEL)`` device pointer, where
-``GFP_KERNEL`` is the kernel allocation flag to use. For the
-scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the device
-pointer (see the `Non-Contiguous Buffers`_
-section).
+bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type.
+You can pass NULL to the device pointer in that case, which is the
+default mode implying to allocate with ``GFP_KRENEL`` flag.
+If you need a different GFP flag, you can pass it by encoding the flag
+into the device pointer via a special macro
+:c:func:`snd_dma_continuous_data()`.
+For the scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the
+device pointer (see the `Non-Contiguous Buffers`_ section).
 
 Once the buffer is pre-allocated, you can use the allocator in the
 ``hw_params`` callback:
@@ -3539,6 +3595,25 @@ Once the buffer is pre-allocated, you can use the allocator in the
 
 Note that you have to pre-allocate to use this function.
 
+Most of drivers use, though, rather the newly introduced "managed
+buffer allocation mode" instead of the manual allocation or release.
+This is done by calling :c:func:`snd_pcm_set_managed_buffer_all()`
+instead of :c:func:`snd_pcm_lib_preallocate_pages_for_all()`.
+
+::
+
+  snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
+                                 &pci->dev, size, max);
+
+where passed arguments are identical in both functions.
+The difference in the managed mode is that PCM core will call
+:c:func:`snd_pcm_lib_malloc_pages()` internally already before calling
+the PCM ``hw_params`` callback, and call :c:func:`snd_pcm_lib_free_pages()`
+after the PCM ``hw_free`` callback automatically.  So the driver
+doesn't have to call these functions explicitly in its callback any
+longer.  This made many driver code having NULL ``hw_params`` and
+``hw_free`` entries.
+
 External Hardware Buffers
 -------------------------
 
@@ -3693,20 +3768,26 @@ provides an interface for handling SG-buffers. The API is provided in
 ``<sound/pcm.h>``.
 
 For creating the SG-buffer handler, call
-:c:func:`snd_pcm_lib_preallocate_pages()` or
-:c:func:`snd_pcm_lib_preallocate_pages_for_all()` with
+:c:func:`snd_pcm_set_managed_buffer()` or
+:c:func:`snd_pcm_set_managed_buffer_all()` with
 ``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI
-pre-allocator. You need to pass ``snd_dma_pci_data(pci)``, where pci is
+pre-allocator. You need to pass ``&pci->dev``, where pci is
 the :c:type:`struct pci_dev <pci_dev>` pointer of the chip as
-well. The ``struct snd_sg_buf`` instance is created as
-``substream->dma_private``. You can cast the pointer like:
+well.
+
+::
+
+  snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+                                 &pci->dev, size, max);
+
+The ``struct snd_sg_buf`` instance is created as
+``substream->dma_private`` in turn. You can cast the pointer like:
 
 ::
 
   struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private;
 
-Then call :c:func:`snd_pcm_lib_malloc_pages()` in the ``hw_params``
-callback as well as in the case of normal PCI buffer. The SG-buffer
+Then in :c:func:`snd_pcm_lib_malloc_pages()` call, the common SG-buffer
 handler will allocate the non-contiguous kernel pages of the given size
 and map them onto the virtually contiguous memory. The virtual pointer
 is addressed in runtime->dma_area. The physical address
@@ -3715,41 +3796,40 @@ physically non-contiguous. The physical address table is set up in
 ``sgbuf->table``. You can get the physical address at a certain offset
 via :c:func:`snd_pcm_sgbuf_get_addr()`.
 
-When a SG-handler is used, you need to set
-:c:func:`snd_pcm_sgbuf_ops_page()` as the ``page`` callback. (See
-`page callback`_ section.)
-
-To release the data, call :c:func:`snd_pcm_lib_free_pages()` in
-the ``hw_free`` callback as usual.
+If you need to release the SG-buffer data explicitly, call the
+standard API function :c:func:`snd_pcm_lib_free_pages()` as usual.
 
 Vmalloc'ed Buffers
 ------------------
 
 It's possible to use a buffer allocated via :c:func:`vmalloc()`, for
-example, for an intermediate buffer. Since the allocated pages are not
-contiguous, you need to set the ``page`` callback to obtain the physical
-address at every offset.
+example, for an intermediate buffer. In the recent version of kernel,
+you can simply allocate it via standard
+:c:func:`snd_pcm_lib_malloc_pages()` and co after setting up the
+buffer preallocation with ``SNDRV_DMA_TYPE_VMALLOC`` type.
 
-The easiest way to achieve it would be to use
-:c:func:`snd_pcm_lib_alloc_vmalloc_buffer()` for allocating the buffer
-via :c:func:`vmalloc()`, and set :c:func:`snd_pcm_sgbuf_ops_page()` to
-the ``page`` callback.  At release, you need to call
-:c:func:`snd_pcm_lib_free_vmalloc_buffer()`.
+::
 
-If you want to implementation the ``page`` manually, it would be like
-this:
+  snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC,
+                                 NULL, 0, 0);
 
-::
+The NULL is passed to the device pointer argument, which indicates
+that the default pages (GFP_KERNEL and GFP_HIGHMEM) will be
+allocated.
 
-  #include <linux/vmalloc.h>
+Also, note that zero is passed to both the size and the max size
+arguments here.  Since each vmalloc call should succeed at any time,
+we don't need to pre-allocate the buffers like other continuous
+pages.
 
-  /* get the physical page pointer on the given offset */
-  static struct page *mychip_page(struct snd_pcm_substream *substream,
-                                  unsigned long offset)
-  {
-          void *pageptr = substream->runtime->dma_area + offset;
-          return vmalloc_to_page(pageptr);
-  }
+If you need the 32bit DMA allocation, pass the device pointer encoded
+by :c:func:`snd_dma_continuous_data()` with ``GFP_KERNEL|__GFP_DMA32``
+argument.
+
+::
+
+  snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC,
+          snd_dma_continuous_data(GFP_KERNEL | __GFP_DMA32), 0, 0);
 
 Proc Interface
 ==============