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
  ...

25 files changed, 540 insertions, 258 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>;
+	};
diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst
index 368a07a165f5..7d7c191102a7 100644
--- a/Documentation/sound/hd-audio/models.rst
+++ b/Documentation/sound/hd-audio/models.rst
@@ -254,10 +254,12 @@ alc274-dell-aio
     ALC274 fixups on Dell AIO machines
 alc255-dummy-lineout
     Dell Precision 3930 fixups
-alc255-dell-headset"},
+alc255-dell-headset
     Dell Precision 3630 fixups
 alc295-hp-x360
     HP Spectre X360 fixups
+alc-sense-combo
+    Headset button support for Chrome platform
 
 ALC66x/67x/892
 ==============
diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
index b37234afdfa1..6b154dbb02cc 100644
--- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
+++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
@@ -3520,14 +3520,14 @@ allocator will try to get an area as large as possible within the
 given size.
 
 The second argument (type) and the third argument (device pointer) are
-dependent on the bus. In the case of the ISA bus, pass
-:c:func:`snd_dma_isa_data()` as the third argument with
+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 PCI
-scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with
-``snd_dma_pci_data(pci)`` (see the `Non-Contiguous Buffers`_
+``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).
 
 Once the buffer is pre-allocated, you can use the allocator in the
@@ -3924,15 +3924,12 @@ The scheme of the real suspend job is as follows.
 2. Call :c:func:`snd_power_change_state()` with
    ``SNDRV_CTL_POWER_D3hot`` to change the power status.
 
-3. Call :c:func:`snd_pcm_suspend_all()` to suspend the running
-   PCM streams.
-
-4. If AC97 codecs are used, call :c:func:`snd_ac97_suspend()` for
+3. If AC97 codecs are used, call :c:func:`snd_ac97_suspend()` for
    each codec.
 
-5. Save the register values if necessary.
+4. Save the register values if necessary.
 
-6. Stop the hardware if necessary.
+5. Stop the hardware if necessary.
 
 A typical code would be like:
 
@@ -3946,12 +3943,10 @@ A typical code would be like:
           /* (2) */
           snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
           /* (3) */
-          snd_pcm_suspend_all(chip->pcm);
-          /* (4) */
           snd_ac97_suspend(chip->ac97);
-          /* (5) */
+          /* (4) */
           snd_mychip_save_registers(chip);
-          /* (6) */
+          /* (5) */
           snd_mychip_stop_hardware(chip);
           return 0;
   }
@@ -3994,13 +3989,9 @@ A typical code would be like:
           return 0;
   }
 
-As shown in the above, it's better to save registers after suspending
-the PCM operations via :c:func:`snd_pcm_suspend_all()` or
-:c:func:`snd_pcm_suspend()`. It means that the PCM streams are
-already stopped when the register snapshot is taken. But, remember that
-you don't have to restart the PCM stream in the resume callback. It'll
-be restarted via trigger call with ``SNDRV_PCM_TRIGGER_RESUME`` when
-necessary.
+Note that, at the time this callback gets called, the PCM stream has
+been already suspended via its own PM ops calling
+:c:func:`snd_pcm_suspend_all()` internally.
 
 OK, we have all callbacks now. Let's set them up. In the initialization
 of the card, make sure that you can get the chip data from the card
diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst
index f6845b2278ea..77f67ded53de 100644
--- a/Documentation/sound/soc/dpcm.rst
+++ b/Documentation/sound/soc/dpcm.rst
@@ -13,7 +13,7 @@ drivers that expose several ALSA PCMs and can route to multiple DAIs.
 The DPCM runtime routing is determined by the ALSA mixer settings in the same
 way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
 graph representing the DSP internal audio paths and uses the mixer settings to
-determine the patch used by each ALSA PCM.
+determine the path used by each ALSA PCM.
 
 DPCM re-uses all the existing component codec, platform and DAI drivers without
 any modifications.
@@ -101,7 +101,7 @@ The audio driver processes this as follows :-
 
 4. Machine driver or audio HAL enables the speaker path.
 
-5. DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
+5. DPCM runs the PCM ops for startup(), hw_params(), prepare() and
    trigger(start) for DAI1 Speakers since the path is enabled.
 
 In this example, the machine driver or userspace audio HAL can alter the routing
@@ -221,7 +221,7 @@ like a BT phone call :-
 This allows the host CPU to sleep while the DSP, MODEM DAI and the BT DAI are
 still in operation.
 
-A BE DAI link can also set the codec to a dummy device if the code is a device
+A BE DAI link can also set the codec to a dummy device if the codec is a device
 that is managed externally.
 
 Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
@@ -249,7 +249,7 @@ configuration.
 	struct snd_interval *channels = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
 
-	/* The DSP will covert the FE rate to 48k, stereo */
+	/* The DSP will convert the FE rate to 48k, stereo */
 	rate->min = rate->max = 48000;
 	channels->min = channels->max = 2;
 
@@ -386,5 +386,3 @@ This means creating a new FE that is connected with a virtual path to both
 DAI links. The DAI links will be started when the FE PCM is started and stopped
 when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
 this configuration.
-
-