Merge tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We had again a busy development cycle with many new drivers as well as
  lots of core improvements / cleanups. Let's go for highlights:

  ALSA core:

   - PCM locking scheme was refactored for reducing a global rwlock

   - PCM suspend is handled in the device type PM ops now; lots of
     explicit calls were reduced by this action

   - Cleanups about PCM buffer preallocation calls

   - Kill NULL device object in memory allocations

   - Lots of procfs API cleanups

  ASoC core:

   - Support for only powering up channels that are actively being used

   - Cleanups / fixes of topology API

  ASoC drivers:

   - MediaTek BTCVSD for a Bluetooth radio chip, which is the first such
     driver we've had upstream!

   - Quite a few improvements to simplify the generic card drivers,
     especially the merge of the SCU cards into the main generic drivers

   - Lots of fixes for probing on Intel systems to follow more standard
     styles

   - A big refresh and cleanup of the Samsung drivers

   - New drivers: Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341
     and CS35L26, Google ChromeOS embedded controllers, Ingenic JZ4725B,
     MediaTek BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328,
     Spreadtrum DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM
     formatters

  ALSA drivers:

   - Improvements of Tegra HD-audio controller driver for supporting new
     chips

   - HD-audio codec quirks for ALC294 S4 resume, ASUS laptop, Chrome
     headset button support and Dell workstations

   - Improved DSD support on USB-audio

   - Quirk for MOTU MicroBook II USB-audio

   - Support for Fireface UCX support and Solid State Logic Duende
     Classic/Mini"

* tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (461 commits)
  ALSA: usb-audio: Add quirk for MOTU MicroBook II
  ASoC: stm32: i2s: skip useless write in slave mode
  ASoC: stm32: i2s: fix race condition in irq handler
  ASoC: stm32: i2s: remove useless callback
  ASoC: stm32: i2s: fix dma configuration
  ASoC: stm32: i2s: fix stream count management
  ASoC: stm32: i2s: fix 16 bit format support
  ASoC: stm32: i2s: fix IRQ clearing
  ASoC: qcom: Kconfig: fix dependency for sdm845
  ASoC: Intel: Boards: Add Maxim98373 support
  ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
  ALSA: firewire-motu: fix construction of PCM frame for capture direction
  ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against Liquid Saffire 56
  ALSA: hda: Extend i915 component bind timeout
  ASoC: wm_adsp: Improve logging messages
  ASoC: wm_adsp: Add support for multiple compressed buffers
  ASoC: wm_adsp: Refactor compress stream initialisation
  ASoC: wm_adsp: Reorder some functions for improved clarity
  ASoC: wm_adsp: Factor out stripping padding from ADSP data
  ASoC: cs35l36: Fix an IS_ERR() vs NULL checking bug
  ...

22 files changed, 520 insertions, 229 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
index e79aeef73f28..9225472c80b4 100644
--- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt
+++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
@@ -17,12 +17,18 @@ Required properties:
                 Documentation/devicetree/bindings/regulator/regulator.txt
 
 Optional properties:
- - reset-gpio:  the reset pin for the chip, for more details consult
+ - reset-gpios: the reset pin for the chip, for more details consult
                 Documentation/devicetree/bindings/gpio/gpio.txt
 
  - DVDD-supply: supply voltage for the digital core, please consult
                 Documentation/devicetree/bindings/regulator/regulator.txt
 
+- adi,micbias: configures the voltage setting for the MICBIAS pin.
+		Select 0/1/2/3/4/5/6/7/8 to specify MICBIAS voltage
+		5V/5.5V/6V/6.5V/7V/7.5V/8V/8.5V/9V
+		If not specified the default value will be "7" meaning 8.5 Volts.
+		This property is only valid for the ADAU1977
+
 For required properties on SPI, please consult
 Documentation/devicetree/bindings/spi/spi-bus.txt
 
@@ -40,7 +46,8 @@ Examples:
 		AVDD-supply = <&regulator>;
 		DVDD-supply = <&regulator_digital>;
 
-		reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+		adi,micbias = <3>;
+		reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
 	};
 
 	adau1977_i2c: adau1977@11 {
@@ -50,5 +57,5 @@ Examples:
 		AVDD-supply = <&regulator>;
 		DVDD-supply = <&regulator_digital>;
 
-		reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+		reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
 	};
diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt
index 7839be78448d..e5820235e0d5 100644
--- a/Documentation/devicetree/bindings/sound/ak4458.txt
+++ b/Documentation/devicetree/bindings/sound/ak4458.txt
@@ -4,7 +4,7 @@ This device supports I2C mode.
 
 Required properties:
 
-- compatible : "asahi-kasei,ak4458"
+- compatible : "asahi-kasei,ak4458" or "asahi-kasei,ak4497"
 - reg : The I2C address of the device for I2C
 
 Optional properties:
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt
deleted file mode 100644
index 62d42768a00b..000000000000
--- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt
+++ /dev/null
@@ -1,123 +0,0 @@
-Audio-Graph-SCU-Card:
-
-Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM".
-
-It is based on common bindings for device graphs.
-see ${LINUX}/Documentation/devicetree/bindings/graph.txt
-
-Basically, Audio-Graph-SCU-Card property is same as
-Simple-Card / Simple-SCU-Card / Audio-Graph-Card.
-see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt
-    ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt
-    ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt
-
-Below are same as Simple-Card / Audio-Graph-Card.
-
-- label
-- dai-format
-- frame-master
-- bitclock-master
-- bitclock-inversion
-- frame-inversion
-- dai-tdm-slot-num
-- dai-tdm-slot-width
-- clocks / system-clock-frequency
-
-Below are same as Simple-SCU-Card.
-
-- convert-rate
-- convert-channels
-- prefix
-- routing
-
-Required properties:
-
-- compatible				: "audio-graph-scu-card";
-- dais					: list of CPU DAI port{s}
-
-Example 1. Sampling Rate Conversion
-
-	sound_card {
-		compatible = "audio-graph-scu-card";
-
-		label = "sound-card";
-		prefix = "codec";
-		routing = "codec Playback", "DAI0 Playback",
-			  "DAI0 Capture",   "codec Capture";
-		convert-rate = <48000>;
-
-		dais = <&cpu_port>;
-	};
-
-	audio-codec {
-		...
-
-		port {
-			codec_endpoint: endpoint {
-				remote-endpoint = <&cpu_endpoint>;
-			};
-		};
-	};
-
-	dai-controller {
-		...
-		cpu_port: port {
-			cpu_endpoint: endpoint {
-				remote-endpoint = <&codec_endpoint>;
-
-				dai-format = "left_j";
-				...
-			};
-		};
-	};
-
-Example 2. 2 CPU 1 Codec (Mixing)
-
-	sound_card {
-		compatible = "audio-graph-scu-card";
-
-		label = "sound-card";
-		routing = "codec Playback", "DAI0 Playback",
-			  "codec Playback", "DAI1 Playback",
-			  "DAI0 Capture",   "codec Capture";
-
-		dais = <&cpu_port0
-			&cpu_port1>;
-	};
-
-	audio-codec {
-		...
-
-		audio-graph-card,prefix = "codec";
-		audio-graph-card,convert-rate = <48000>;
-		port {
-			codec_endpoint0: endpoint {
-				remote-endpoint = <&cpu_endpoint0>;
-			};
-			codec_endpoint1: endpoint {
-				remote-endpoint = <&cpu_endpoint1>;
-			};
-		};
-	};
-
-	dai-controller {
-		...
-		ports {
-			cpu_port0: port {
-				cpu_endpoint0: endpoint {
-					remote-endpoint = <&codec_endpoint0>;
-
-					dai-format = "left_j";
-					...
-				};
-			};
-			cpu_port1: port {
-				cpu_endpoint1: endpoint {
-					remote-endpoint = <&codec_endpoint1>;
-
-					dai-format = "left_j";
-					...
-				};
-			};
-		};
-	};
diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt
new file mode 100644
index 000000000000..912bd162b477
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l36.txt
@@ -0,0 +1,168 @@
+CS35L36 Speaker Amplifier
+
+Required properties:
+
+  - compatible : "cirrus,cs35l36"
+
+  - reg : the I2C address of the device for I2C
+
+  - VA-supply, VP-supply : power supplies for the device,
+  as covered in
+  Documentation/devicetree/bindings/regulator/regulator.txt.
+
+  - cirrus,boost-ctl-millivolt : Boost Voltage Value.  Configures the boost
+  converter's output voltage in mV. The range is from 2550mV to 12000mV with
+  increments of 50mV.
+  (Default) VP
+
+  - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA.
+  Configures the peak current by monitoring the current through the boost FET.
+  Range starts at 1600mA and goes to a maximum of 4500mA with increments of
+  50mA.
+  (Default) 4.50 Amps
+
+  - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value.
+  Seeds the digital boost converter's inductor estimation block with the initial
+  inductance value to reference.
+
+  1000 = 1uH (Default)
+  1200 = 1.2uH
+
+Optional properties:
+  - cirrus,multi-amp-mode : Boolean to determine if there are more than
+  one amplifier in the system. If more than one it is best to Hi-Z the ASP
+  port to prevent bus contention on the output signal
+
+  - cirrus,boost-ctl-select : Boost conerter control source selection.
+  Selects the source of the BST_CTL target VBST voltage for the boost
+  converter to generate.
+  0x00 - Control Port Value
+  0x01 - Class H Tracking (Default)
+  0x10 - MultiDevice Sync Value
+
+  - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming
+  PCM data
+
+  - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the
+  polarity of outbound IMON feedback data
+
+  - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the
+  polarity of outbound VMON feedback data
+
+  - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable.
+  This enables the digital boost converter to operate in a low power
+  (Discontinuous Conduction) mode during low loading conditions.
+
+  - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is
+  reduced when operating in a Weak-FET Drive Mode and must not be used to drive
+  a large load.
+
+  - cirrus,classh-wk-fet-delay :  Weak-FET entry delay. Controls the delay
+  (in ms) before the Class H algorithm switches to the weak-FET voltage
+  (after the audio falls and remains below the value specified in WKFET_AMP_THLD).
+
+  0 = 0ms
+  1 = 5ms
+  2 = 10ms
+  3 = 50ms
+  4 = 100ms (Default)
+  5 = 200ms
+  6 = 500ms
+  7 = 1000ms
+
+  - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold.
+  Configures the signal threshold at which the PWM output stage enters
+  weak-FET operation. The range is 50mV to 700mV in 50mV increments.
+
+  - cirrus,temp-warn-threshold :  Amplifier overtemperature warning threshold.
+  Configures the threshold at which the overtemperature warning condition occurs.
+  When the threshold is met, the overtemperature warning attenuation is applied
+  and the TEMP_WARN_EINT interrupt status bit is set.
+  If TEMP_WARN_MASK = 0, INTb is asserted.
+
+  0 = 105C
+  1 = 115C
+  2 = 125C (Default)
+  3 = 135C
+
+  - cirrus,irq-drive-select : Selects the driver type of the selected interrupt
+  output.
+
+  0 = Open-drain
+  1 = Push-pull (Default)
+
+  - cirrus,irq-gpio-select : Selects the pin to serve as the programmable
+  interrupt output.
+
+  0 = PDM_DATA / SWIRE_SD / INT (Default)
+  1 = GPIO
+
+Optional properties for the "cirrus,vpbr-config" Sub-node
+
+  - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the
+  VBST brownout prevention algorithm is enabled or disabled.
+
+  0 = VBST brownout prevention disabled (default)
+  1 = VBST brownout prevention enabled
+
+  See Section 7.31.1 VPBR Config for configuration options & further details
+
+  - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout
+  threshold voltage
+
+  - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the
+  amount delay between consecutive volume attenuation steps when a brownout
+  condition is present and the VP brownout condition is in an attacking state.
+
+  - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP
+  brownout prevention attacking attenuation step size when operating in either
+  digital volume or analog gain modes.
+
+  - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention
+  can apply to the audio signal.
+
+  - cirrus,vpbr-wait : Configures the delay time between a brownout condition
+  no longer being present and the VP brownout prevention entering an attenuation
+  release state.
+
+  - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay
+  between consecutive volume attenuation release steps when a brownout condition
+  is not longer present and the VP brownout is in an attenuation release state.
+
+  - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value
+  is reached, the error condition still remains, and this bit is set, the audio
+  is muted.
+
+Example:
+
+cs35l36: cs35l36@40 {
+	compatible = "cirrus,cs35l36";
+	reg = <0x40>;
+	VA-supply = <&dummy_vreg>;
+	VP-supply = <&dummy_vreg>;
+	reset-gpios = <&gpio0 54 0>;
+	interrupt-parent = <&gpio8>;
+	interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+
+	cirrus,boost-ind-nanohenry = <1000>;
+	cirrus,boost-ctl-millivolt = <10000>;
+	cirrus,boost-peak-milliamp = <4500>;
+	cirrus,boost-ctl-select = <0x00>;
+	cirrus,weak-fet-delay = <0x04>;
+	cirrus,weak-fet-thld = <0x01>;
+	cirrus,temp-warn-threshold = <0x01>;
+	cirrus,multi-amp-mode;
+	cirrus,irq-drive-select = <0x01>;
+	cirrus,irq-gpio-select = <0x01>;
+
+	cirrus,vpbr-config {
+		cirrus,vpbr-en = <0x00>;
+		cirrus,vpbr-thld = <0x05>;
+		cirrus,vpbr-atk-rate = <0x02>;
+		cirrus,vpbr-atk-vol = <0x01>;
+		cirrus,vpbr-max-attn = <0x09>;
+		cirrus,vpbr-wait = <0x01>;
+		cirrus,vpbr-rel-rate = <0x05>;
+		cirrus,vpbr-mute-en = <0x00>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt
new file mode 100644
index 000000000000..12b4aa8ef0db
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4341.txt
@@ -0,0 +1,22 @@
+Cirrus Logic CS4341 audio DAC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+  - compatible: "cirrus,cs4341a"
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+For required properties on I2C-bus, please consult
+Documentation/devicetree/bindings/i2c/i2c.txt
+For required properties on SPI-bus, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Example:
+	codec: cs4341@0 {
+		#sound-dai-cells = <0>;
+		compatible = "cirrus,cs4341a";
+		reg = <0>;
+		spi-max-frequency = <6000000>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
index b279b6072bd5..a58f79f5345c 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -45,6 +45,23 @@ Optional properties:
 - fck_parent : Should contain a valid clock name which will be used as parent
 	       for the McASP fck
 
+Optional GPIO support:
+If any McASP pin need to be used as GPIO then the McASP node must have:
+...
+  gpio-controller
+  #gpio-cells = <2>;
+...
+
+When requesting a GPIO, the first parameter is the PIN index in McASP_P*
+registers.
+For example to request the AXR2 pin of mcasp8:
+function-gpios = <&mcasp8 2 0>;
+
+Or to request the ACLKR pin of mcasp8:
+function-gpios = <&mcasp8 29 0>;
+
+For generic gpio information, please refer to bindings/gpio/gpio.txt
+
 Example:
 
 mcasp0: mcasp0@1d00000 {
diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.txt b/Documentation/devicetree/bindings/sound/fsl,micfil.txt
new file mode 100644
index 000000000000..53e227b15277
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,micfil.txt
@@ -0,0 +1,32 @@
+NXP MICFIL Digital Audio Interface (MICFIL).
+
+The MICFIL digital interface provides a 16-bit audio signal from a PDM
+microphone bitstream in a configurable output sampling rate.
+
+Required properties:
+
+  - compatible		: Compatible list, contains "fsl,imx8mm-micfil"
+
+  - reg			: Offset and length of the register set for the device.
+
+  - interrupts		: Contains the micfil interrupts.
+
+  - clocks		: Must contain an entry for each entry in clock-names.
+
+  - clock-names		: Must include the "ipg_clk" for register access and
+			  "ipg_clk_app" for internal micfil clock.
+
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+Example:
+micfil: micfil@30080000 {
+	compatible = "fsl,imx8mm-micfil";
+	reg = <0x0 0x30080000 0x0 0x10000>;
+	interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>,
+		     <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&clk IMX8MM_CLK_PDM_IPG>,
+		 <&clk IMX8MM_CLK_PDM_ROOT>;
+	clock-names = "ipg_clk", "ipg_clk_app";
+	dmas = <&sdma2 24 26 0x80000000>;
+};
diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt
new file mode 100644
index 000000000000..1084f7f22eea
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt
@@ -0,0 +1,26 @@
+* 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.
+
+An EC codec node should only be found as a sub-node of the EC node (see
+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.
+
+Example:
+
+cros-ec@0 {
+	compatible = "google,cros-ec-spi";
+
+	...
+
+	cros_ec_codec: ec-codec {
+		compatible = "google,cros-ec-codec";
+		#sound-dai-cells = <1>;
+		max-dmic-gain = <43>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt
new file mode 100644
index 000000000000..05adc0d47b13
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt
@@ -0,0 +1,20 @@
+Ingenic JZ4725B codec controller
+
+Required properties:
+- compatible : "ingenic,jz4725b-codec"
+- reg : codec registers location and length
+- clocks : phandle to the AIC clock.
+- clock-names: must be set to "aic".
+- #sound-dai-cells: Must be set to 0.
+
+Example:
+
+codec: audio-codec@100200a4 {
+	compatible = "ingenic,jz4725b-codec";
+	reg = <0x100200a4 0x8>;
+
+	#sound-dai-cells = <0>;
+
+	clocks = <&cgu JZ4725B_CLK_AIC>;
+	clock-names = "aic";
+};
diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt
new file mode 100644
index 000000000000..1ffcade87e7b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt
@@ -0,0 +1,20 @@
+Ingenic JZ4740 codec controller
+
+Required properties:
+- compatible : "ingenic,jz4740-codec"
+- reg : codec registers location and length
+- clocks : phandle to the AIC clock.
+- clock-names: must be set to "aic".
+- #sound-dai-cells: Must be set to 0.
+
+Example:
+
+codec: audio-codec@10020080 {
+	compatible = "ingenic,jz4740-codec";
+	reg = <0x10020080 0x8>;
+
+	#sound-dai-cells = <0>;
+
+	clocks = <&cgu JZ4740_CLK_AIC>;
+	clock-names = "aic";
+};
diff --git a/Documentation/devicetree/bindings/sound/mt6358.txt b/Documentation/devicetree/bindings/sound/mt6358.txt
new file mode 100644
index 000000000000..5465730013a1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt6358.txt
@@ -0,0 +1,18 @@
+Mediatek MT6358 Audio Codec
+
+The communication between MT6358 and SoC is through Mediatek PMIC wrapper.
+For more detail, please visit Mediatek PMIC wrapper documentation.
+
+Must be a child node of PMIC wrapper.
+
+Required properties:
+
+- compatible : "mediatek,mt6358-sound".
+- Avdd-supply : power source of AVDD
+
+Example:
+
+mt6358_snd {
+	compatible = "mediatek,mt6358-sound";
+	Avdd-supply = <&mt6358_vaud28_reg>;
+};
diff --git a/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt
new file mode 100644
index 000000000000..396ba38619f6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt
@@ -0,0 +1,36 @@
+Mediatek AFE PCM controller for mt8183
+
+Required properties:
+- compatible = "mediatek,mt68183-audio";
+- reg: register location and size
+- interrupts: should contain AFE interrupt
+- power-domains: should define the power domain
+- clocks: Must contain an entry for each entry in clock-names
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"mtkaif_26m_clk",
+		"top_mux_audio",
+		"top_mux_aud_intbus",
+		"top_sys_pll3_d4",
+		"top_clk26m_clk";
+
+Example:
+
+	afe: mt8183-afe-pcm@11220000  {
+		compatible = "mediatek,mt8183-audio";
+		reg = <0 0x11220000 0 0x1000>;
+		interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>;
+		power-domains = <&scpsys MT8183_POWER_DOMAIN_AUDIO>;
+		clocks = <&infrasys CLK_INFRA_AUDIO>,
+			 <&infrasys CLK_INFRA_AUDIO_26M_BCLK>,
+			 <&topckgen CLK_TOP_MUX_AUDIO>,
+			 <&topckgen CLK_TOP_MUX_AUD_INTBUS>,
+			 <&topckgen CLK_TOP_SYSPLL_D2_D4>,
+			 <&clk26m>;
+		clock-names = "infra_sys_audio_clk",
+			      "mtkaif_26m_clk",
+			      "top_mux_audio",
+			      "top_mux_aud_intbus",
+			      "top_sys_pll_d2_d4",
+			      "top_clk26m_clk";
+	};
diff --git a/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt
new file mode 100644
index 000000000000..679e44839b48
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt
@@ -0,0 +1,24 @@
+Mediatek ALSA BT SCO CVSD/MSBC Driver
+
+Required properties:
+- compatible = "mediatek,mtk-btcvsd-snd";
+- reg: register location and size of PKV and SRAM_BANK2
+- interrupts: should contain BTSCO interrupt
+- mediatek,infracfg: the phandles of INFRASYS
+- mediatek,offset: Array contains of register offset and mask
+    infra_misc_offset,
+    infra_conn_bt_cvsd_mask,
+    cvsd_mcu_read_offset,
+    cvsd_mcu_write_offset,
+    cvsd_packet_indicator_offset
+
+Example:
+
+	mtk-btcvsd-snd@18000000 {
+		compatible = "mediatek,mtk-btcvsd-snd";
+		reg=<0 0x18000000 0 0x1000>,
+		    <0 0x18080000 0 0x8000>;
+		interrupts = <GIC_SPI 286 IRQ_TYPE_LEVEL_LOW>;
+		mediatek,infracfg = <&infrasys>;
+		mediatek,offset = <0xf00 0x800 0xfd0 0xfd4 0xfd8>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
index 44d27456e8a4..21cd310963b1 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
@@ -13,6 +13,10 @@ Required properties:
   See ../reset/reset.txt for details.
 - reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x
 
+Optional properties:
+- nvidia,model : The user-visible name of this sound complex. Since the property
+  is optional, legacy boards can use default name provided in hda driver.
+
 Example:
 
 hda@70030000 {
@@ -27,4 +31,5 @@ hda@70030000 {
 		 <&tegra_car 128>, /* hda2hdmi */
 		 <&tegra_car 111>; /* hda2codec_2x */
 	reset-names = "hda", "hda2hdmi", "hda2codec_2x";
+	nvidia,model = "jetson-tk1-hda";
 };
diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
index fdcea3d12ee5..e7d17dda55db 100644
--- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
@@ -30,6 +30,7 @@ Required properties
  - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node.
  - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node.
  - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node.
+
 Optional Properties:
  - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons
 			     detection on headset when the mbhc is powered up
@@ -92,9 +93,9 @@ spmi_bus {
 				  "cdc_ear_cnp_int",
 				  "cdc_hphr_cnp_int",
 				  "cdc_hphl_cnp_int";
-	               VDD-CDC-IO-supply = <&pm8916_l5>;
-	               VDD-CDC-TX-RX-CX-supply = <&pm8916_l5>;
-	               VDD-MICBIAS-supply = <&pm8916_l13>;
+	               vdd-cdc-io-supply = <&pm8916_l5>;
+	               vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>;
+	               vdd-micbias-supply = <&pm8916_l13>;
 	               #sound-dai-cells = <1>;
 	};
 };
diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
index 1d8d49e30af7..5d6ea66a863f 100644
--- a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
@@ -34,12 +34,12 @@ Required properties with SLIMbus Interface:
 	Definition: Interrupt names of WCD INTR1 and INTR2
 	Should be: "intr1", "intr2"
 
-- reset-gpio:
+- reset-gpios:
 	Usage: required
 	Value type: <String Array>
 	Definition: Reset gpio line
 
-- qcom,ifd:
+- slim-ifc-dev:
 	Usage: required
 	Value type: <phandle>
 	Definition: SLIM interface device
@@ -104,13 +104,13 @@ Required properties with SLIMbus Interface:
 	Value type: <u32>
 	Definition: Must be 1
 
-codec@1{
+audio-codec@1{
 	compatible = "slim217,1a0";
 	reg  = <1 0>;
 	interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>;
 	interrupt-names = "intr2"
-	reset-gpio = <&msmgpio 64 0>;
-	qcom,ifd  = <&wc9335_ifd>;
+	reset-gpios = <&msmgpio 64 0>;
+	slim-ifc-dev  = <&wc9335_ifd>;
 	clock-names = "mclk", "native";
 	clocks = <&rpmcc RPM_SMD_DIV_CLK1>,
 		 <&rpmcc RPM_SMD_BB_CLK1>;
diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt
new file mode 100644
index 000000000000..2469588c7ccb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt
@@ -0,0 +1,23 @@
+* Rockchip Rk3328 internal codec
+
+Required properties:
+
+- compatible: "rockchip,rk3328-codec"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- rockchip,grf: the phandle of the syscon node for GRF register.
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should be "pclk".
+- spk-depop-time-ms: speak depop time msec.
+
+Example for rk3328 internal codec:
+
+codec: codec@ff410000 {
+	compatible = "rockchip,rk3328-codec";
+	reg = <0x0 0xff410000 0x0 0x1000>;
+	rockchip,grf = <&grf>;
+	clocks = <&cru PCLK_ACODEC>;
+	clock-names = "pclk";
+	spk-depop-time-ms = 100;
+	status = "disabled";
+};
diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
index 9c58f724396a..9d9ff5184939 100644
--- a/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -37,6 +37,15 @@ VDDIO		1.8V		2.5V		3.3V
 2 =		3.33 mA		5.74 mA		8.03  mA
 3 =		4.99 mA		8.61 mA		12.05 mA
 
+- sclk-strength: the SCLK pad strength. Possible values are:
+0, 1, 2 and 3 as per the table below:
+
+VDDIO		1.8V		2.5V		3.3V
+0 = 		Disable
+1 =		1.66 mA		2.87 mA		4.02  mA
+2 =		3.33 mA		5.74 mA		8.03  mA
+3 =		4.99 mA		8.61 mA		12.05 mA
+
 Example:
 
 sgtl5000: codec@a {
diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt
deleted file mode 100644
index 3a2f71616cda..000000000000
--- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt
+++ /dev/null
@@ -1,94 +0,0 @@
-ASoC Simple SCU Sound Card
-
-Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM".
-For example, you can use this driver if you want to exchange sampling rate convert,
-Mixing, etc...
-
-Required properties:
-
-- compatible				: "simple-scu-audio-card"
-					  "renesas,rsrc-card"
-Optional properties:
-
-- simple-audio-card,name		: see simple-audio-card.txt
-- simple-audio-card,cpu			: see simple-audio-card.txt
-- simple-audio-card,codec		: see simple-audio-card.txt
-
-Optional subnode properties:
-
-- simple-audio-card,format		: see simple-audio-card.txt
-- simple-audio-card,frame-master	: see simple-audio-card.txt
-- simple-audio-card,bitclock-master	: see simple-audio-card.txt
-- simple-audio-card,bitclock-inversion	: see simple-audio-card.txt
-- simple-audio-card,frame-inversion	: see simple-audio-card.txt
-- simple-audio-card,convert-rate	: platform specified sampling rate convert
-- simple-audio-card,convert-channels	: platform specified converted channel size (2 - 8 ch)
-- simple-audio-card,prefix		: see routing
-- simple-audio-card,widgets		: Please refer to widgets.txt.
-- simple-audio-card,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.
-					  use audio-prefix if some components is using same sink/sources naming.
-					  it can be used if compatible was "renesas,rsrc-card";
-
-Required CPU/CODEC subnodes properties:
-
-- sound-dai				: see simple-audio-card.txt
-
-Optional CPU/CODEC subnodes properties:
-
-- clocks / system-clock-frequency	: see simple-audio-card.txt
-
-Example 1. Sampling Rate Conversion
-
-sound {
-	compatible = "simple-scu-audio-card";
-
-	simple-audio-card,name = "rsnd-ak4643";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&sndcodec>;
-	simple-audio-card,frame-master = <&sndcodec>;
-
-	simple-audio-card,convert-rate = <48000>;
-
-	simple-audio-card,prefix = "ak4642";
-	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
-			"DAI0 Capture", "ak4642 Capture";
-
-	sndcpu: simple-audio-card,cpu {
-		sound-dai = <&rcar_sound>;
-	};
-
-	sndcodec: simple-audio-card,codec {
-		sound-dai = <&ak4643>;
-		system-clock-frequency = <11289600>;
-	};
-};
-
-Example 2. 2 CPU 1 Codec (Mixing)
-
-sound {
-	compatible = "simple-scu-audio-card";
-
-	simple-audio-card,name = "rsnd-ak4643";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&dpcmcpu>;
-	simple-audio-card,frame-master = <&dpcmcpu>;
-
-	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
-			"ak4642 Playback", "DAI1 Playback";
-
-	dpcmcpu: cpu@0 {
-		sound-dai = <&rcar_sound 0>;
-	};
-
-	cpu@1 {
-		sound-dai = <&rcar_sound 1>;
-	};
-
-	codec {
-		prefix = "ak4642";
-		sound-dai = <&ak4643>;
-		clocks = <&audio_clock>;
-	};
-};
diff --git a/Documentation/devicetree/bindings/sound/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt
new file mode 100644
index 000000000000..4b23e84b2e57
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sprd-pcm.txt
@@ -0,0 +1,23 @@
+* Spreadtrum DMA platfrom bindings
+
+Required properties:
+- compatible: Should be "sprd,pcm-platform".
+- dmas: Specify the 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.
+
+Example:
+
+	audio_platform:platform@0 {
+		compatible = "sprd,pcm-platform";
+		dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>,
+		     <&agcp_dma 3 3>, <&agcp_dma 4 4>,
+		     <&agcp_dma 5 5>, <&agcp_dma 6 6>,
+		     <&agcp_dma 7 7>, <&agcp_dma 8 8>,
+		     <&agcp_dma 9 9>, <&agcp_dma 10 10>;
+		dma-names = "normal_p_l", "normal_p_r",
+			"normal_c_l", "normal_c_r",
+			"voice_c", "fast_p",
+			"loop_c", "loop_p",
+			"voip_c", "voip_p";
+	};
diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt
new file mode 100644
index 000000000000..cbc93c8f4963
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt
@@ -0,0 +1,29 @@
+Device-Tree bindings for Xilinx PL audio formatter
+
+The IP core supports DMA, data formatting(AES<->PCM conversion)
+of audio samples.
+
+Required properties:
+ - compatible: "xlnx,audio-formatter-1.0"
+ - interrupt-names: Names specified to list of interrupts in same
+		    order mentioned under "interrupts".
+		    List of supported interrupt names are:
+		    "irq_mm2s" : interrupt from MM2S block
+		    "irq_s2mm" : interrupt from S2MM block
+ - interrupts-parent: Phandle for interrupt controller.
+ - interrupts: List of Interrupt numbers.
+ - reg: Base address and size of the IP core instance.
+ - clock-names: List of input clocks.
+   Required elements: "s_axi_lite_aclk", "aud_mclk"
+ - clocks: Input clock specifier. Refer to common clock bindings.
+
+Example:
+	audio_ss_0_audio_formatter_0: audio_formatter@80010000 {
+		compatible = "xlnx,audio-formatter-1.0";
+		interrupt-names = "irq_mm2s", "irq_s2mm";
+		interrupt-parent = <&gic>;
+		interrupts = <0 104 4>, <0 105 4>;
+		reg = <0x0 0x80010000 0x0 0x1000>;
+		clock-names = "s_axi_lite_aclk", "aud_mclk";
+		clocks = <&clk 71>, <&clk_wiz_1 0>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/xlnx,spdif.txt b/Documentation/devicetree/bindings/sound/xlnx,spdif.txt
new file mode 100644
index 000000000000..15c2d64d247c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/xlnx,spdif.txt
@@ -0,0 +1,28 @@
+Device-Tree bindings for Xilinx SPDIF IP
+
+The IP supports playback and capture of SPDIF audio
+
+Required properties:
+ - compatible: "xlnx,spdif-2.0"
+ - clock-names: List of input clocks.
+   Required elements: "s_axi_aclk", "aud_clk_i"
+ - clocks: Input clock specifier. Refer to common clock bindings.
+ - reg: Base address and address length of the IP core instance.
+ - interrupts-parent: Phandle for interrupt controller.
+ - interrupts: List of Interrupt numbers.
+ - xlnx,spdif-mode: 0 :- receiver mode
+		    1 :- transmitter mode
+ - xlnx,aud_clk_i: input audio clock value.
+
+Example:
+	spdif_0: spdif@80010000 {
+		clock-names = "aud_clk_i", "s_axi_aclk";
+		clocks = <&misc_clk_0>, <&clk 71>;
+		compatible = "xlnx,spdif-2.0";
+		interrupt-names = "spdif_interrupt";
+		interrupt-parent = <&gic>;
+		interrupts = <0 91 4>;
+		reg = <0x0 0x80010000 0x0 0x10000>;
+		xlnx,spdif-mode = <1>;
+		xlnx,aud_clk_i = <49152913>;
+	};