Merge tag 'sound-4.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "The majority of this update is about ASoC, including a few new
  drivers, and the rest are mostly minor changes.  The only substantial
  change in ALSA core is about the additional error handling in the
  compress-offload API.  Below are highlights:

   - Add the error propagating support in compress-offload API

   - HD-audio: a usual Dell headset fixup, an Intel HDMI/DP fix, and the
     default mixer setup change ot turn off the loopback

   - Lots of updates for ASoC Intel drivers, mostly board support and
     bug fixing, and to the NAU8825 driver

   - Work on generalizing bits of simple-card to allow more code sharing
     with the Renesas rsrc-card (which can't use simple-card due to DPCM)

   - Removal of the Odroid X2 driver due to replacement with simple-card

   - Support for several new Mediatek platforms and associated boards

   - New ASoC drivers for Allwinner A10, Analog Devices ADAU7002,
     Broadcom Cygnus, Cirrus Logic CS35L33 and CS53L30, Maxim MAX8960
     and MAX98504, Realtek RT5514 and Wolfson WM8758"

* tag 'sound-4.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (278 commits)
  sound: oss: Use kernel_read_file_from_path() for mod_firmware_load()
  ASoC: Intel: Skylake: Delete an unnecessary check before the function call "release_firmware"
  ASoC: Intel: Skylake: Fix NULL Pointer exception in dynamic_debug.
  ASoC: samsung: Specify DMA channels through struct snd_dmaengine_pcm_config
  ASoC: samsung: Fix error paths in the I2S driver's probe()
  ASoC: cs53l30: Fix bit shift issue of TDM mode
  ASoC: cs53l30: Fix a bug for TDM slot location validation
  ASoC: rockchip: correct the spdif clk
  ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members
  ASoC: rsrc-card: use asoc_simple_card_parse_card_name()
  ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai
  ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name()
  ASoC: simple-card: use asoc_simple_card_parse_card_name()
  ASoC: simple-card-utils: add asoc_simple_card_parse_card_name()
  ASoC: simple-card: use asoc_simple_card_parse_dailink_name()
  ASoC: simple-card-utils: add asoc_simple_card_set_dailink_name()
  ASoC: nau8825: drop redundant idiom when converting integer to boolean
  ASoC: nau8825: jack connection decision with different insertion logic
  ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver
  ASoC: mediatek: mt2701: fix non static symbol warning
  ...

23 files changed, 625 insertions, 65 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
index 8dbce0e18dda..1447dec28125 100644
--- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
+++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
@@ -13,6 +13,11 @@ Required properties:
  - reg:			The i2c address. Value depends on the state of ADDR0
 			and ADDR1, as wired in hardware.
 
+Optional properties:
+ - clock-names:		If provided must be "mclk".
+ - clocks:		phandle + clock-specifiers for the clock that provides
+			the audio master clock for the device.
+
 Examples:
 #include <dt-bindings/sound/adau17x1.h>
 
@@ -20,5 +25,8 @@ Examples:
 		adau1361@38 {
 			compatible = "adi,adau1761";
 			reg = <0x38>;
+
+			clock-names = "mclk";
+			clocks = <&audio_clock>;
 		};
 	};
diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt
new file mode 100644
index 000000000000..f144ee1abf85
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau7002.txt
@@ -0,0 +1,19 @@
+Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter
+
+Required properties:
+
+ - compatible: Must be "adi,adau7002"
+
+Optional properties:
+
+ - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD
+	supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+	If this property is not present it is assumed that the supply pin is
+	hardwired to always on.
+
+Example:
+	adau7002: pdm-to-i2s {
+		compatible = "adi,adau7002";
+		IOVDD-supply = <&supply>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
new file mode 100644
index 000000000000..b139e66d2a11
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
@@ -0,0 +1,67 @@
+BROADCOM Cygnus Audio I2S/TDM/SPDIF controller
+
+Required properties:
+	- compatible : "brcm,cygnus-audio"
+	- #address-cells: 32bit valued, 1 cell.
+	- #size-cells:  32bit valued, 0 cell.
+	- reg : Should contain audio registers location and length
+	- reg-names: names of the registers listed in "reg" property
+		Valid names are "aud" and "i2s_in". "aud" contains a
+		set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains
+		a set of I2S_IN registers.
+	- clocks: PLL and leaf clocks used by audio ports
+	- assigned-clocks: PLL and leaf clocks
+	- assigned-clock-parents: parent clocks of the assigned clocks
+		(usually the PLL)
+	- assigned-clock-rates: List of clock frequencies of the
+		assigned clocks
+	- clock-names: names of 3 leaf clocks used by audio ports
+		Valid names are "ch0_audio", "ch1_audio", "ch2_audio"
+	- interrupts: audio DMA interrupt number
+
+SSP Subnode properties:
+- reg: The index of ssp port interface to use
+	Valid value are 0, 1, 2, or 3 (for spdif)
+
+Example:
+	cygnus_audio: audio@180ae000 {
+		compatible = "brcm,cygnus-audio";
+		#address-cells = <1>;
+		#size-cells = <0>;
+		reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>;
+		reg-names = "aud", "i2s_in";
+		clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>,
+				<&audiopll BCM_CYGNUS_AUDIOPLL_CH1>,
+				<&audiopll BCM_CYGNUS_AUDIOPLL_CH2>;
+		assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>,
+							<&audiopll BCM_CYGNUS_AUDIOPLL_CH0>,
+							<&audiopll BCM_CYGNUS_AUDIOPLL_CH1>,
+							<&audiopll BCM_CYGNUS_AUDIOPLL_CH2>;
+		assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>;
+		assigned-clock-rates = <1769470191>,
+								<0>,
+								<0>,
+								<0>;
+		clock-names = "ch0_audio", "ch1_audio", "ch2_audio";
+		interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>;
+
+		ssp0: ssp_port@0 {
+			reg = <0>;
+			status = "okay";
+		};
+
+		ssp1: ssp_port@1 {
+			reg = <1>;
+			status = "disabled";
+		};
+
+		ssp2: ssp_port@2 {
+			reg = <2>;
+			status = "disabled";
+		};
+
+		spdif: spdif_port@3 {
+			reg = <3>;
+			status = "disabled";
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/bt-sco.txt b/Documentation/devicetree/bindings/sound/bt-sco.txt
index 29b8e5d40203..641edf75e184 100644
--- a/Documentation/devicetree/bindings/sound/bt-sco.txt
+++ b/Documentation/devicetree/bindings/sound/bt-sco.txt
@@ -4,7 +4,7 @@ This device support generic Bluetooth SCO link.
 
 Required properties:
 
-  - compatible : "delta,dfbmcs320"
+  - compatible : "delta,dfbmcs320" or "linux,bt-sco"
 
 Example:
 
diff --git a/Documentation/devicetree/bindings/sound/cs35l33.txt b/Documentation/devicetree/bindings/sound/cs35l33.txt
new file mode 100644
index 000000000000..acfb47525b49
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l33.txt
@@ -0,0 +1,126 @@
+CS35L33 Speaker Amplifier
+
+Required properties:
+
+  - compatible : "cirrus,cs35l33"
+
+  - 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.
+
+Optional properties:
+
+  - reset-gpios : gpio used to reset the amplifier
+
+  - interrupt-parent : Specifies the phandle of the interrupt controller to
+    which the IRQs from CS35L33 are delivered to.
+ -  interrupts : IRQ line info CS35L33.
+    (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
+    for further information relating to interrupt properties)
+
+  - cirrus,boost-ctl : Booster voltage use to supply the amp. If the value is
+    0, then VBST = VP. If greater than 0, the boost voltage will be 3300mV with
+    a value of 1 and will increase at a step size of 100mV until a maximum of
+    8000mV.
+
+  - cirrus,ramp-rate : On power up, it affects the time from when the power
+    up sequence begins to the time the audio reaches a full-scale output.
+    On power down, it affects the time from when the power-down sequence
+    begins to when the amplifier disables the PWM outputs. If this property
+    is not set then soft ramping will be disabled and ramp time would be
+    20ms. If this property is set to 0,1,2,3 then ramp times would be 40ms,
+    60ms,100ms,175ms respectively for 48KHz sample rate.
+
+  - cirrus,boost-ipk : The maximum current allowed for the boost converter.
+    The range starts at 1850000uA and goes to a maximum of 3600000uA
+    with a step size of 15625uA. The default is 2500000uA.
+
+  - cirrus,imon-adc-scale : Configures the scaling of data bits from the IMON
+    ADC data word. This property can be set as a value of 0 for bits 15 down
+    to 0, 6 for 21 down to 6, 7, for 22 down to 7, 8 for 23 down to 8.
+
+
+Optional H/G Algorithm sub-node:
+
+The cs35l33 node can have a single "cirrus,hg-algo" sub-node that will enable
+the internal H/G Algorithm.
+
+  - cirrus,hg-algo : Sub-node for internal Class H/G algorithm that
+    controls the amplifier supplies.
+
+Optional properties for the "cirrus,hg-algo" sub-node:
+
+  - cirrus,mem-depth : Memory depth for the Class H/G algorithm measured in
+    LRCLK cycles. If this property is set to 0, 1, 2, or 3 then the memory
+    depths will be 1, 4, 8, 16 LRCLK cycles.  The default is 16 LRCLK cycles.
+
+    cirrus,release-rate : The number of consecutive LRCLK periods before
+    allowing release condition tracking updates. The number of LRCLK periods
+    start at 3 to a maximum of 255.
+
+  - cirrus,ldo-thld : Configures the signal threshold at which the PWM output
+    stage enters LDO operation. Starts as a default value of 50mV for a value
+    of 1 and increases with a step size of 50mV to a maximum of 750mV (value of
+    0xF).
+
+  - cirrus,ldo-path-disable : This is a boolean property. If present, the H/G
+    algorithm uses the max detection path.  If not present, the LDO
+    detection path is used.
+
+  - cirrus,ldo-entry-delay : The LDO entry delay in milliseconds before the H/G
+    algorithm switches to the LDO voltage.  This property can be set to values
+    from 0 to 7 for delays of 5ms, 10ms, 50ms, 100ms, 200ms, 500ms, 1000ms.
+    The default is 100ms.
+
+  - cirrus,vp-hg-auto : This is a boolean property.  When set, class H/G VPhg
+    automatic updating is enabled.
+
+  - cirrus,vp-hg :  Class H/G algorithm VPhg.  Controls the H/G algorithm's
+    reference to the VP voltage for when to start generating a boosted VBST.
+    The reference voltage starts at 3000mV with a value of 0x3 and is increased
+    by 100mV per step to a maximum of 5500mV.
+
+  - cirrus,vp-hg-rate : The rate (number of LRCLK periods) at which the VPhg is
+    allowed to increase to a higher voltage when using VPhg automatic
+    tracking. This property can be set to values from 0 to 3 with rates of 128
+    periods, 2048 periods, 32768 periods, and 524288 periods.
+    The default is 32768 periods.
+
+  - cirrus,vp-hg-va : VA calculation reference for automatic VPhg tracking
+    using VPMON. This property can be set to values from 0 to 6 starting at
+    1800mV with a step size of 50mV up to a maximum value of 1750mV.
+    Default is 1800mV.
+
+Example:
+
+cs35l33: cs35l33@40 {
+	compatible = "cirrus,cs35l33";
+	reg = <0x40>;
+
+	VA-supply = <&ldo5_reg>;
+	VP-supply = <&ldo5_reg>;
+
+	interrupt-parent = <&gpio8>;
+	interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+
+	reset-gpios = <&cs47l91 34 0>;
+
+	cirrus,ramp-rate = <0x0>;
+	cirrus,boost-ctl = <0x30>;  /* VBST = 8000mV */
+	cirrus,boost-ipk = <0xE0>; /* 3600mA */
+	cirrus,imon-adc-scale = <0> /* Bits 15 down to 0 */
+
+	cirrus,hg-algo {
+		cirrus,mem-depth = <0x3>;
+		cirrus,release-rate = <0x3>;
+		cirrus,ldo-thld = <0x1>;
+		cirrus,ldo-path-disable = <0x0>;
+		cirrus,ldo-entry-delay=<0x4>;
+		cirrus,vp-hg-auto;
+		cirrus,vp-hg=<0xF>;
+		cirrus,vp-hg-rate=<0x2>;
+		cirrus,vp-hg-va=<0x0>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt
new file mode 100644
index 000000000000..4dbfb8274cd9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs53l30.txt
@@ -0,0 +1,44 @@
+CS53L30 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs53l30"
+
+  - reg : the I2C address of the device
+
+  - VA-supply, VP-supply : power supplies for the device,
+    as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin.
+
+  - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either
+		 GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled
+		 by the driver automatically.
+
+  - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin.
+			 0 = Hi-Z
+			 1 = 1.80 V
+			 2 = 2.75 V
+
+  - cirrus,use-sdout2 : This is a boolean property. If present, it indicates
+			the hardware design connects both SDOUT1 and SDOUT2
+			pins to output data. Otherwise, it indicates that
+			only SDOUT1 is connected for data output.
+			* CS53l30 supports 4-channel data output in the same
+			* frame using two different ways:
+			* 1) Normal I2S mode on two data pins -- each SDOUT
+			*    carries 2-channel data in the same time.
+			* 2) TDM mode on one signle data pin -- SDOUT1 carries
+			*    4-channel data per frame.
+
+Example:
+
+codec: cs53l30@48 {
+	compatible = "cirrus,cs53l30";
+	reg = <0x48>;
+	reset-gpios = <&gpio 54 0>;
+	VA-supply = <&cs53l30_va>;
+	VP-supply = <&cs53l30_vp>;
+};
diff --git a/Documentation/devicetree/bindings/sound/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt
index 7bb54247f8e8..6a536d570e29 100644
--- a/Documentation/devicetree/bindings/sound/designware-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/designware-i2s.txt
@@ -12,6 +12,10 @@ Required properties:
    one for receive.
  - dma-names : "tx" for the transmit channel, "rx" for the receive channel.
 
+Optional properties:
+ - interrupts: The interrupt line number for the I2S controller. Add this
+   parameter if the I2S controller that you are using does not support DMA.
+
 For more details on the 'dma', 'dma-names', 'clock' and 'clock-names'
 properties please check:
 	* resource-names.txt
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index ceaef5126989..f749e2744824 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -58,7 +58,7 @@ Required properties:
 			   * DMIC (stands for Digital Microphone Jack)
 
 			  Note: The "Mic Jack" and "AMIC" are redundant while
-			        coexsiting in order to support the old bindings
+			        coexisting in order to support the old bindings
 				of wm8962 and sgtl5000.
 
 Optional properties:
diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt
new file mode 100644
index 000000000000..583ed5fdfb28
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98504.txt
@@ -0,0 +1,44 @@
+Maxim MAX98504 class D mono speaker amplifier
+
+This device supports I2C control interface and an IRQ output signal. It features
+a PCM and PDM digital audio interface (DAI) and a differential analog input.
+
+Required properties:
+
+ - compatible : "maxim,max98504"
+ - reg : should contain the I2C slave device address
+ - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device,
+   as covered in ../regulator/regulator.txt
+ - interrupts : should specify the interrupt line the device is connected to,
+   as described in ../interrupt-controller/interrupts.txt
+
+Optional properties:
+
+ - maxim,brownout-threshold - the PVDD brownout threshold, the value must be
+   from 0, 1...21 range, corresponding to 2.6V, 2.65V...3.65V voltage range
+ - maxim,brownout-attenuation - the brownout attenuation to the speaker gain
+   applied during the "attack hold" and "timed hold" phase, the value must be
+   from 0...6 (dB) range
+ - maxim,brownout-attack-hold-ms - the brownout attack hold phase time in ms,
+   0...255 (VBATBROWN_ATTK_HOLD, register 0x0018)
+ - maxim,brownout-timed-hold-ms - the brownout timed hold phase time in ms,
+   0...255 (VBATBROWN_TIME_HOLD, register 0x0019)
+ - maxim,brownout-release-rate-ms - the brownout release phase step time in ms,
+   0...255 (VBATBROWN_RELEASE, register 0x001A)
+
+The default value when the above properties are not specified is 0,
+the maxim,brownout-threshold property must be specified to actually enable
+the PVDD brownout protection.
+
+Example:
+
+ max98504@31 {
+	compatible = "maxim,max98504";
+	reg = <0x31>;
+	interrupt-parent = <&gpio_bank_0>;
+	interrupts = <2 0>;
+
+	DVDD-supply = <&regulator>;
+	DIOVDD-supply = <&regulator>;
+	PVDD-supply = <&regulator>;
+};
diff --git a/Documentation/devicetree/bindings/sound/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt
new file mode 100644
index 000000000000..e0d4e95e31b3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max9860.txt
@@ -0,0 +1,28 @@
+MAX9860 Mono Audio Voice Codec
+
+Required properties:
+
+  - compatible : "maxim,max9860"
+
+  - reg : the I2C address of the device
+
+  - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for
+    the device, as covered in bindings/regulator/regulator.txt
+
+  - clock-names : Required element: "mclk".
+
+  - clocks : A clock specifier for the clock connected as MCLK.
+
+Examples:
+
+	max9860: max9860@10 {
+		compatible = "maxim,max9860";
+		reg = <0x10>;
+
+		AVDD-supply = <&reg_1v8>;
+		DVDD-supply = <&reg_1v8>;
+		DVDDIO-supply = <&reg_3v0>;
+
+		clock-names = "mclk";
+		clocks = <&pck2>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
new file mode 100644
index 000000000000..3e623a724e55
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
@@ -0,0 +1,150 @@
+Mediatek AFE PCM controller for mt2701
+
+Required properties:
+- compatible = "mediatek,mt2701-audio";
+- reg: register location and size
+- interrupts: Should contain AFE interrupt
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"top_audio_mux1_sel",
+		"top_audio_mux2_sel",
+		"top_audio_mux1_div",
+		"top_audio_mux2_div",
+		"top_audio_48k_timing",
+		"top_audio_44k_timing",
+		"top_audpll_mux_sel",
+		"top_apll_sel",
+		"top_aud1_pll_98M",
+		"top_aud2_pll_90M",
+		"top_hadds2_pll_98M",
+		"top_hadds2_pll_294M",
+		"top_audpll",
+		"top_audpll_d4",
+		"top_audpll_d8",
+		"top_audpll_d16",
+		"top_audpll_d24",
+		"top_audintbus_sel",
+		"clk_26m",
+		"top_syspll1_d4",
+		"top_aud_k1_src_sel",
+		"top_aud_k2_src_sel",
+		"top_aud_k3_src_sel",
+		"top_aud_k4_src_sel",
+		"top_aud_k5_src_sel",
+		"top_aud_k6_src_sel",
+		"top_aud_k1_src_div",
+		"top_aud_k2_src_div",
+		"top_aud_k3_src_div",
+		"top_aud_k4_src_div",
+		"top_aud_k5_src_div",
+		"top_aud_k6_src_div",
+		"top_aud_i2s1_mclk",
+		"top_aud_i2s2_mclk",
+		"top_aud_i2s3_mclk",
+		"top_aud_i2s4_mclk",
+		"top_aud_i2s5_mclk",
+		"top_aud_i2s6_mclk",
+		"top_asm_m_sel",
+		"top_asm_h_sel",
+		"top_univpll2_d4",
+		"top_univpll2_d2",
+		"top_syspll_d5";
+
+Example:
+
+	afe: mt2701-afe-pcm@11220000 {
+		compatible = "mediatek,mt2701-audio";
+		reg = <0 0x11220000 0 0x2000>,
+		      <0 0x112A0000 0 0x20000>;
+		interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>,
+			     <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>;
+		clocks = <&infracfg CLK_INFRA_AUDIO>,
+			 <&topckgen CLK_TOP_AUD_MUX1_SEL>,
+			 <&topckgen CLK_TOP_AUD_MUX2_SEL>,
+			 <&topckgen CLK_TOP_AUD_MUX1_DIV>,
+			 <&topckgen CLK_TOP_AUD_MUX2_DIV>,
+			 <&topckgen CLK_TOP_AUD_48K_TIMING>,
+			 <&topckgen CLK_TOP_AUD_44K_TIMING>,
+			 <&topckgen CLK_TOP_AUDPLL_MUX_SEL>,
+			 <&topckgen CLK_TOP_APLL_SEL>,
+			 <&topckgen CLK_TOP_AUD1PLL_98M>,
+			 <&topckgen CLK_TOP_AUD2PLL_90M>,
+			 <&topckgen CLK_TOP_HADDS2PLL_98M>,
+			 <&topckgen CLK_TOP_HADDS2PLL_294M>,
+			 <&topckgen CLK_TOP_AUDPLL>,
+			 <&topckgen CLK_TOP_AUDPLL_D4>,
+			 <&topckgen CLK_TOP_AUDPLL_D8>,
+			 <&topckgen CLK_TOP_AUDPLL_D16>,
+			 <&topckgen CLK_TOP_AUDPLL_D24>,
+			 <&topckgen CLK_TOP_AUDINTBUS_SEL>,
+			 <&clk26m>,
+			 <&topckgen CLK_TOP_SYSPLL1_D4>,
+			 <&topckgen CLK_TOP_AUD_K1_SRC_SEL>,
+			 <&topckgen CLK_TOP_AUD_K2_SRC_SEL>,
+			 <&topckgen CLK_TOP_AUD_K3_SRC_SEL>,
+			 <&topckgen CLK_TOP_AUD_K4_SRC_SEL>,
+			 <&topckgen CLK_TOP_AUD_K5_SRC_SEL>,
+			 <&topckgen CLK_TOP_AUD_K6_SRC_SEL>,
+			 <&topckgen CLK_TOP_AUD_K1_SRC_DIV>,
+			 <&topckgen CLK_TOP_AUD_K2_SRC_DIV>,
+			 <&topckgen CLK_TOP_AUD_K3_SRC_DIV>,
+			 <&topckgen CLK_TOP_AUD_K4_SRC_DIV>,
+			 <&topckgen CLK_TOP_AUD_K5_SRC_DIV>,
+			 <&topckgen CLK_TOP_AUD_K6_SRC_DIV>,
+			 <&topckgen CLK_TOP_AUD_I2S1_MCLK>,
+			 <&topckgen CLK_TOP_AUD_I2S2_MCLK>,
+			 <&topckgen CLK_TOP_AUD_I2S3_MCLK>,
+			 <&topckgen CLK_TOP_AUD_I2S4_MCLK>,
+			 <&topckgen CLK_TOP_AUD_I2S5_MCLK>,
+			 <&topckgen CLK_TOP_AUD_I2S6_MCLK>,
+			 <&topckgen CLK_TOP_ASM_M_SEL>,
+			 <&topckgen CLK_TOP_ASM_H_SEL>,
+			 <&topckgen CLK_TOP_UNIVPLL2_D4>,
+			 <&topckgen CLK_TOP_UNIVPLL2_D2>,
+			 <&topckgen CLK_TOP_SYSPLL_D5>;
+
+		clock-names = "infra_sys_audio_clk",
+			      "top_audio_mux1_sel",
+			      "top_audio_mux2_sel",
+			      "top_audio_mux1_div",
+			      "top_audio_mux2_div",
+			      "top_audio_48k_timing",
+			      "top_audio_44k_timing",
+			      "top_audpll_mux_sel",
+			      "top_apll_sel",
+			      "top_aud1_pll_98M",
+			      "top_aud2_pll_90M",
+			      "top_hadds2_pll_98M",
+			      "top_hadds2_pll_294M",
+			      "top_audpll",
+			      "top_audpll_d4",
+			      "top_audpll_d8",
+			      "top_audpll_d16",
+			      "top_audpll_d24",
+			      "top_audintbus_sel",
+			      "clk_26m",
+			      "top_syspll1_d4",
+			      "top_aud_k1_src_sel",
+			      "top_aud_k2_src_sel",
+			      "top_aud_k3_src_sel",
+			      "top_aud_k4_src_sel",
+			      "top_aud_k5_src_sel",
+			      "top_aud_k6_src_sel",
+			      "top_aud_k1_src_div",
+			      "top_aud_k2_src_div",
+			      "top_aud_k3_src_div",
+			      "top_aud_k4_src_div",
+			      "top_aud_k5_src_div",
+			      "top_aud_k6_src_div",
+			      "top_aud_i2s1_mclk",
+			      "top_aud_i2s2_mclk",
+			      "top_aud_i2s3_mclk",
+			      "top_aud_i2s4_mclk",
+			      "top_aud_i2s5_mclk",
+			      "top_aud_i2s6_mclk",
+			      "top_asm_m_sel",
+			      "top_asm_h_sel",
+			      "top_univpll2_d4",
+			      "top_univpll2_d2",
+			      "top_syspll_d5";
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt
new file mode 100644
index 000000000000..05574446ceb6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt
@@ -0,0 +1,43 @@
+MT2701 with CS42448 CODEC
+
+Required properties:
+- compatible: "mediatek,mt2701-cs42448-machine"
+- mediatek,platform: the phandle of MT2701 ASoC platform
+- audio-routing: a list of the connections between audio
+- mediatek,audio-codec: the phandles of cs42448 codec
+- mediatek,audio-codec-bt-mrg the phandles of bt-sco dummy codec
+- pinctrl-names: Should contain only one value - "default"
+- pinctrl-0: Should specify pin control groups used for this controller.
+- i2s1-in-sel-gpio1, i2s1-in-sel-gpio2: Should specify two gpio pins to
+					control I2S1-in mux.
+
+Example:
+
+	sound:sound {
+		compatible = "mediatek,mt2701-cs42448-machine";
+		mediatek,platform = <&afe>;
+		/* CS42448 Machine name */
+		audio-routing =
+			"Line Out Jack", "AOUT1L",
+			"Line Out Jack", "AOUT1R",
+			"Line Out Jack", "AOUT2L",
+			"Line Out Jack", "AOUT2R",
+			"Line Out Jack", "AOUT3L",
+			"Line Out Jack", "AOUT3R",
+			"Line Out Jack", "AOUT4L",
+			"Line Out Jack", "AOUT4R",
+			"AIN1L", "AMIC",
+			"AIN1R", "AMIC",
+			"AIN2L", "Tuner In",
+			"AIN2R", "Tuner In",
+			"AIN3L", "Satellite Tuner In",
+			"AIN3R", "Satellite Tuner In",
+			"AIN3L", "AUX In",
+			"AIN3R", "AUX In";
+		mediatek,audio-codec = <&cs42448>;
+		mediatek,audio-codec-bt-mrg = <&bt_sco_codec>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&aud_pins_default>;
+		i2s1-in-sel-gpio1 = <&pio 53 0>;
+		i2s1-in-sel-gpio2 = <&pio 54 0>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
index 5bfa6b60530b..29dce2ac8773 100644
--- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
@@ -1,8 +1,9 @@
-MT8173 with RT5650 CODECS
+MT8173 with RT5650 CODECS and HDMI via I2S
 
 Required properties:
 - compatible : "mediatek,mt8173-rt5650"
 - mediatek,audio-codec: the phandles of rt5650 codecs
+                        and of the hdmi encoder node
 - mediatek,platform: the phandle of MT8173 ASoC platform
 
 Optional subnodes:
@@ -12,12 +13,17 @@ Required codec-capture subnode properties:
   <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1)
   <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2)
 
+- mediatek,mclk: the MCLK source
+  0 : external oscillator, MCLK = 12.288M
+  1 : internal source from mt8173, MCLK = sampling rate*256
+
 Example:
 
 	sound {
 		compatible = "mediatek,mt8173-rt5650";
-		mediatek,audio-codec = <&rt5650>;
+		mediatek,audio-codec = <&rt5650 &hdmi0>;
 		mediatek,platform = <&afe>;
+		mediatek,mclk = <0>;
 		codec-capture {
 			sound-dai = <&rt5650 1>;
 		};
diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
index 0741dff048dd..6f6c2f8e908d 100644
--- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
+++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
@@ -8,6 +8,8 @@ Required properties:
 - interrupts: Interrupt number for McPDM
 - interrupt-parent: The parent interrupt controller
 - ti,hwmods: Name of the hwmod associated to the McPDM
+- clocks:  phandle for the pdmclk provider, likely <&twl6040>
+- clock-names: Must be "pdmclk"
 
 Example:
 
@@ -19,3 +21,11 @@ mcpdm: mcpdm@40132000 {
 	interrupt-parent = <&gic>;
 	ti,hwmods = "mcpdm";
 };
+
+In board DTS file the pdmclk needs to be added:
+
+&mcpdm {
+	clocks = <&twl6040>;
+	clock-names = "pdmclk";
+	status = "okay";
+};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index c7b29df4a963..15a7316e4c91 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -373,6 +373,8 @@ Optional properties:
 - #clock-cells			: it must be 0 if your system has audio_clkout
 				  it must be 1 if your system has audio_clkout0/1/2/3
 - clock-frequency		: for all audio_clkout0/1/2/3
+- clkout-lr-asynchronous	: boolean property. it indicates that audio_clkoutn
+				  is asynchronizes with lr-clock.
 
 SSI subnode properties:
 - interrupts			: Should contain SSI interrupt for PIO transfer
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
index 6e86d8aa29b4..4ea29aa9af59 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -23,6 +23,11 @@ Required properties:
 - rockchip,playback-channels: max playback channels, if not set, 8 channels default.
 - rockchip,capture-channels: max capture channels, if not set, 2 channels default.
 
+Required properties for controller which support multi channels
+playback/capture:
+
+- rockchip,grf: the phandle of the syscon node for GRF register.
+
 Example for rk3288 I2S controller:
 
 i2s@ff890000 {
diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt
index e24436fc5ea9..9cabfc18cb57 100644
--- a/Documentation/devicetree/bindings/sound/rt5514.txt
+++ b/Documentation/devicetree/bindings/sound/rt5514.txt
@@ -8,6 +8,11 @@ Required properties:
 
 - reg : The I2C address of the device.
 
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+- clock-names: Should be "mclk"
+
 Pins on the device (for linking into audio routes) for RT5514:
 
   * DMIC1L
diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt
deleted file mode 100644
index 9148f72319e1..000000000000
--- a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt
+++ /dev/null
@@ -1,35 +0,0 @@
-Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec
-
-Required properties:
- - compatible : "samsung,odroidx2-audio" - for Odroid X2 board,
-		"samsung,odroidu3-audio" - for Odroid U3 board
- - samsung,model : the user-visible name of this sound complex
- - samsung,i2s-controller : the phandle of the I2S controller
- - samsung,audio-codec : the phandle of the MAX98090 audio codec
- - 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:
-     * Headphone Jack
-     * Speakers
-
-Example:
-
-sound {
-	compatible = "samsung,odroidu3-audio";
-	samsung,i2s-controller = <&i2s0>;
-	samsung,audio-codec = <&max98090>;
-	samsung,model = "Odroid-X2";
-	samsung,audio-routing =
-		"Headphone Jack", "HPL",
-		"Headphone Jack", "HPR",
-		"IN1", "Mic Jack",
-		"Mic Jack", "MICBIAS";
-};
diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
index 0e5e4eb3ef1b..5666da7b8605 100644
--- a/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -7,6 +7,14 @@ Required properties:
 
 - clocks : the clock provider of SYS_MCLK
 
+- VDDA-supply : the regulator provider of VDDA
+
+- VDDIO-supply: the regulator provider of VDDIO
+
+Optional properties:
+
+- VDDD-supply : the regulator provider of VDDD
+
 - micbias-resistor-k-ohms : the bias resistor to be used in kOmhs
 	The resistor can take values of 2k, 4k or 8k.
 	If set to 0 it will be off.
@@ -15,17 +23,9 @@ Required properties:
 
 - micbias-voltage-m-volts : the bias voltage to be used in mVolts
 	The voltage can take values from 1.25V to 3V by 250mV steps
-	If this node is not mentionned or the value is unknown, then
+	If this node is not mentioned or the value is unknown, then
 	the value is set to 1.25V.
 
-- VDDA-supply : the regulator provider of VDDA
-
-- VDDIO-supply: the regulator provider of VDDIO
-
-Optional properties:
-
-- VDDD-supply : the regulator provider of VDDD
-
 Example:
 
 codec: sgtl5000@0a {
diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
index 4d9a83d9a017..16bcdfb6760e 100644
--- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
@@ -33,11 +33,11 @@ Required properties:
 	"tx" for "st,sti-uni-player" compatibility
 	"rx" for "st,sti-uni-reader" compatibility
 
-  - version: IP version integrated in SOC.
+  - st,version: IP version integrated in SOC.
 
   - dai-name: DAI name that describes the IP.
 
-  - IP mode: IP working mode depending on associated codec.
+  - st,mode: IP working mode depending on associated codec.
 	"HDMI" connected to HDMI codec and support IEC HDMI formats (player only).
 	"SPDIF" connected to SPDIF codec and support SPDIF formats (player only).
 	"PCM" PCM standard mode for I2S or TDM bus.
@@ -47,7 +47,7 @@ Required properties ("st,sti-uni-player" compatibility only):
   - clocks: CPU_DAI IP clock source, listed in the same order than the
 	    CPU_DAI properties.
 
-  - uniperiph-id: internal SOC IP instance ID.
+  - st,uniperiph-id: internal SOC IP instance ID.
 
 Optional properties:
   - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for
@@ -84,9 +84,9 @@ Example:
 		dmas = <&fdma0 4 0 1>;
 		dai-name = "Uni Player #2 (DAC)";
 		dma-names = "tx";
-		uniperiph-id = <2>;
-		version = <5>;
-		mode = "PCM";
+		st,uniperiph-id = <2>;
+		st,version = <5>;
+		st,mode = "PCM";
 	};
 
 	sti_uni_player3: sti-uni-player@3 {
@@ -100,9 +100,9 @@ Example:
 		dmas = <&fdma0 7 0 1>;
 		dma-names = "tx";
 		dai-name = "Uni Player #3 (SPDIF)";
-		uniperiph-id = <3>;
-		version = <5>;
-		mode = "SPDIF";
+		st,uniperiph-id = <3>;
+		st,version = <5>;
+		st,mode = "SPDIF";
 	};
 
 	sti_uni_reader1: sti-uni-reader@1 {
@@ -115,7 +115,7 @@ Example:
 		dmas = <&fdma0 6 0 1>;
 		dma-names = "rx";
 		dai-name = "Uni Reader #1 (HDMI RX)";
-		version = <3>;
+		st,version = <3>;
 		st,mode = "PCM";
 	};
 
diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
new file mode 100644
index 000000000000..7b526ec64991
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
@@ -0,0 +1,34 @@
+* Allwinner A10 I2S controller
+
+The I2S bus (Inter-IC sound bus) is a serial link for digital
+audio data transfer between devices in the system.
+
+Required properties:
+
+- compatible: should be one of the followings
+   - "allwinner,sun4i-a10-i2s"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- interrupts: should contain the I2S interrupt.
+- dmas: DMA specifiers 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 followings:
+   - "apb" : clock for the I2S bus interface
+   - "mod" : module clock for the I2S controller
+- #sound-dai-cells : Must be equal to 0
+
+Example:
+
+i2s0: i2s@01c22400 {
+	#sound-dai-cells = <0>;
+	compatible = "allwinner,sun4i-a10-i2s";
+	reg = <0x01c22400 0x400>;
+	interrupts = <GIC_SPI 16 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&apb0_gates 3>, <&i2s0_clk>;
+	clock-names = "apb", "mod";
+	dmas = <&dma SUN4I_DMA_NORMAL 3>,
+	       <&dma SUN4I_DMA_NORMAL 3>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 74056dba52be..6bf2d2063b52 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -3,7 +3,7 @@ ASoC Machine Driver
 
 The ASoC machine (or board) driver is the code that glues together all the
 component drivers (e.g. codecs, platforms and DAIs). It also describes the
-relationships between each componnent which include audio paths, GPIOs,
+relationships between each component which include audio paths, GPIOs,
 interrupts, clocking, jacks and voltage regulators.
 
 The machine driver can contain codec and platform specific code. It registers
diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
index 1b6473f393a8..9d579aefbffd 100644
--- a/Documentation/sound/alsa/timestamping.txt
+++ b/Documentation/sound/alsa/timestamping.txt
@@ -14,7 +14,7 @@ provides a refined estimate with a delay.
 event or application query.
 The difference (tstamp - trigger_tstamp) defines the elapsed time.
 
-The ALSA API provides reports two basic pieces of information, avail
+The ALSA API provides two basic pieces of information, avail
 and delay, which combined with the trigger and current system
 timestamps allow for applications to keep track of the 'fullness' of
 the ring buffer and the amount of queued samples.
@@ -53,21 +53,21 @@ case):
 The analog time is taken at the last stage of the playback, as close
 as possible to the actual transducer
 
-The link time is taken at the output of the SOC/chipset as the samples
+The link time is taken at the output of the SoC/chipset as the samples
 are pushed on a link. The link time can be directly measured if
 supported in hardware by sample counters or wallclocks (e.g. with
 HDAudio 24MHz or PTP clock for networked solutions) or indirectly
 estimated (e.g. with the frame counter in USB).
 
 The DMA time is measured using counters - typically the least reliable
-of all measurements due to the bursty natured of DMA transfers.
+of all measurements due to the bursty nature of DMA transfers.
 
 The app time corresponds to the time tracked by an application after
 writing in the ring buffer.
 
-The application can query what the hardware supports, define which
+The application can query the hardware capabilities, define which
 audio time it wants reported by selecting the relevant settings in
-audio_tstamp_config fields, get an estimate of the timestamp
+audio_tstamp_config fields, thus get an estimate of the timestamp
 accuracy. It can also request the delay-to-analog be included in the
 measurement. Direct access to the link time is very interesting on
 platforms that provide an embedded DSP; measuring directly the link
@@ -169,7 +169,7 @@ playback: systime: 938107562 nsec, audio time 938112708 nsec, 	systime delta -51
 Example 1 shows that the timestamp at the DMA level is close to 1ms
 ahead of the actual playback time (as a side time this sort of
 measurement can help define rewind safeguards). Compensating for the
-DMA-link delay in example 2 helps remove the hardware buffering abut
+DMA-link delay in example 2 helps remove the hardware buffering but
 the information is still very jittery, with up to one sample of
 error. In example 3 where the timestamps are measured with the link
 wallclock, the timestamps show a monotonic behavior and a lower