Merge tag 'asoc-v4.5' into asoc-next

ASoC: Updates for v4.5

 - Some more fixes for the topology code, though it is still not final
   and ready for enabling in production.  We really need to get to the
   point where that can be done.
 - A pile of changes for Intel SkyLake drivers which hopefully deliver
   some useful initial functionality for systems with this chipset,
   though there is more work still to come.
 - New drivers for a number of Imagination Technologies IPs.
 - Lots of new features and cleanups for the Renesas drivers.
 - ANC support for WM5110.
 - New driver for Atmel class D speaker drivers.
 - New drivers for Cirrus CS47L24 and WM1831.
 - New driver for Dialog DA7128.
 - New drivers for Realtek RT5659 and RT56156.
 - New driver for Rockchip RK3036.

# gpg: Signature made Wed 23 Dec 2015 00:36:01 GMT using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: key 00000000 occurs more than once in the trustdb
# gpg: key 16005C11: no public key for trusted key - skipped
# gpg: key 16005C11 marked as ultimately trusted
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

Documentation/devicetree/bindings/sound/ak4613.txt

@@ -7,6 +7,16 @@ Required properties:
 - compatible : "asahi-kasei,ak4613"
 - reg : The chip select number on the I2C bus
 
+Optional properties:
+- asahi-kasei,in1-single-end	: Boolean. Indicate input / output pins are single-ended.
+- asahi-kasei,in2-single-end	  rather than differential.
+- asahi-kasei,out1-single-end
+- asahi-kasei,out2-single-end
+- asahi-kasei,out3-single-end
+- asahi-kasei,out4-single-end
+- asahi-kasei,out5-single-end
+- asahi-kasei,out6-single-end
+
 Example:
 
 &i2c {

Documentation/devicetree/bindings/sound/atmel-classd.txt

@@ -16,6 +16,10 @@ Required properties:
 	Required elements: "pclk", "gclk" and "aclk".
 - clocks
 	Please refer to clock-bindings.txt.
+- assigned-clocks
+	Should be <&classd_gclk>.
+- assigned-clock-parents
+	Should be <&audio_pll_pmc>.
 
 Optional properties:
 - pinctrl-names, pinctrl-0
@@ -43,6 +47,8 @@ classd: classd@fc048000 {
 		dma-names = "tx";
 		clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>;
 		clock-names = "pclk", "gclk", "aclk";
+		assigned-clocks = <&classd_gclk>;
+		assigned-clock-parents = <&audio_pll_pmc>;
 
 		pinctrl-names = "default";
 		pinctrl-0 = <&pinctrl_classd_default>;

Documentation/devicetree/bindings/sound/atmel-pdmic.txt

+* Atmel PDMIC driver under ALSA SoC architecture
+
+Required properties:
+- compatible
+	Should be "atmel,sama5d2-pdmic".
+- reg
+	Should contain PDMIC registers location and length.
+- interrupts
+	Should contain the IRQ line for the PDMIC.
+- dmas
+	One DMA specifiers as described in atmel-dma.txt and dma.txt files.
+- dma-names
+	Must be "rx".
+- clock-names
+	Required elements:
+	- "pclk"	peripheral clock
+	- "gclk"	generated clock
+- clocks
+	Must contain an entry for each required entry in clock-names.
+	Please refer to clock-bindings.txt.
+- atmel,mic-min-freq
+	The minimal frequency that the micphone supports.
+- atmel,mic-max-freq
+	The maximal frequency that the micphone supports.
+
+Optional properties:
+- pinctrl-names, pinctrl-0
+	Please refer to pinctrl-bindings.txt.
+- atmel,model
+	The user-visible name of this sound card.
+	The default value is "PDMIC".
+- atmel,mic-offset
+	The offset that should be added.
+	The range is from -32768 to 32767.
+	The default value is 0.
+
+Example:
+	pdmic@f8018000 {
+				compatible = "atmel,sama5d2-pdmic";
+				reg = <0xf8018000 0x124>;
+				interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>;
+				dmas = <&dma0
+					(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
+					| AT91_XDMAC_DT_PERID(50))>;
+				dma-names = "rx";
+				clocks = <&pdmic_clk>, <&pdmic_gclk>;
+				clock-names = "pclk", "gclk";
+
+				pinctrl-names = "default";
+				pinctrl-0 = <&pinctrl_pdmic_default>;
+				atmel,model = "PDMIC @ sama5d2_xplained";
+				atmel,mic-min-freq = <1000000>;
+				atmel,mic-max-freq = <3246000>;
+				atmel,mic-offset = <0x0>;
+	};

Documentation/devicetree/bindings/sound/da7218.txt

+Dialog Semiconductor DA7218 Audio Codec bindings
+
+DA7218 is an audio codec with HP detect feature.
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7217" or "dlg,da7218"
+- reg: Specifies the I2C slave address
+
+- VDD-supply: VDD power supply for the device
+- VDDMIC-supply: VDDMIC power supply for the device
+- VDDIO-supply: VDDIO power supply for the device
+  (See Documentation/devicetree/bindings/regulator/regulator.txt for further
+   information relating to regulators)
+
+Optional properties:
+- interrupt-parent: Specifies the phandle of the interrupt controller to which
+  the IRQs from DA7218 are delivered to.
+- interrupts: IRQ line info for DA7218 chip.
+  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
+   further information relating to interrupt properties)
+- interrupt-names : Name associated with interrupt line. Should be "wakeup" if
+  interrupt is to be used to wake system, otherwise "irq" should be used.
+- wakeup-source: Flag to indicate this device can wake system (suspend/resume).
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias1-lvl-millivolt : Voltage (mV) for Mic Bias 1
+	[<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>]
+- dlg,micbias2-lvl-millivolt : Voltage (mV) for Mic Bias 2
+	[<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>]
+- dlg,mic1-amp-in-sel : Mic1 input source type
+	["diff", "se_p", "se_n"]
+- dlg,mic2-amp-in-sel : Mic2 input source type
+	["diff", "se_p", "se_n"]
+- dlg,dmic1-data-sel : DMIC1 channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic1-samplephase : When to sample audio from DMIC1.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic1-clkrate-hz : DMic1 clock frequency (Hz).
+	[<1500000>, <3000000>]
+- dlg,dmic2-data-sel : DMic2 channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic2-samplephase : When to sample audio from DMic2.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic2-clkrate-hz : DMic2 clock frequency (Hz).
+	[<1500000>, <3000000>]
+- dlg,hp-diff-single-supply : Boolean flag, use single supply for HP
+			      (DA7217 only)
+
+======
+
+Optional Child node - 'da7218_hpldet' (DA7218 only):
+
+Optional properties:
+- dlg,jack-rate-us : Time between jack detect measurements (us)
+	[<5>, <10>, <20>, <40>, <80>, <160>, <320>, <640>]
+- dlg,jack-debounce : Number of debounce measurements taken for jack detect
+	[<0>, <2>, <3>, <4>]
+- dlg,jack-threshold-pct : Threshold level for jack detection (% of VDD)
+	[<84>, <88>, <92>, <96>]
+- dlg,comp-inv : Boolean flag, invert comparator output
+- dlg,hyst : Boolean flag, enable hysteresis
+- dlg,discharge : Boolean flag, auto discharge of Mic Bias on jack removal
+
+======
+
+Example:
+
+	codec: da7218@1a {
+		compatible = "dlg,da7218";
+		reg = <0x1a>;
+		interrupt-parent = <&gpio6>;
+		interrupts = <11 IRQ_TYPE_LEVEL_HIGH>;
+		wakeup-source;
+
+		VDD-supply = <&reg_audio>;
+		VDDMIC-supply = <&reg_audio>;
+		VDDIO-supply = <&reg_audio>;
+
+		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,micbias1-lvl-millivolt = <2600>;
+		dlg,micbias2-lvl-millivolt = <2600>;
+		dlg,mic1-amp-in-sel = "diff";
+		dlg,mic2-amp-in-sel = "diff";
+
+		dlg,dmic1-data-sel = "lrise_rfall";
+		dlg,dmic1-samplephase = "on_clkedge";
+		dlg,dmic1-clkrate-hz = <3000000>;
+		dlg,dmic2-data-sel = "lrise_rfall";
+		dlg,dmic2-samplephase = "on_clkedge";
+		dlg,dmic2-clkrate-hz = <3000000>;
+
+		da7218_hpldet {
+			dlg,jack-rate-us = <40>;
+			dlg,jack-debounce = <2>;
+			dlg,jack-threshold-pct = <84>;
+			dlg,hyst;
+		};
+	};

Documentation/devicetree/bindings/sound/da7219.txt

@@ -28,13 +28,15 @@ Optional properties:
 - clocks : phandle and clock specifier for codec MCLK.
 - clock-names : Clock name string for 'clocks' attribute, should be "mclk".
 
-- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine
-	[<1050>, <1100>, <1200>, <1400>]
 - dlg,micbias-lvl : Voltage (mV) for Mic Bias
-	[<1800>, <2000>, <2200>, <2400>, <2600>]
+	[<1600>, <1800>, <2000>, <2200>, <2400>, <2600>]
 - dlg,mic-amp-in-sel : Mic input source type
 	["diff", "se_p", "se_n"]
 
+Deprecated properties:
+- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine
+  (LDO unavailable in production HW so property no longer required).
+
 ======
 
 Child node - 'da7219_aad':

Documentation/devicetree/bindings/sound/fsl,asrc.txt

@@ -25,6 +25,11 @@ Required properties:
 	"mem"		  Peripheral access clock to access registers.
 	"ipg"		  Peripheral clock to driver module.
 	"asrck_<0-f>"	  Clock sources for input and output clock.
+	"spba"		  The spba clock is required when ASRC is placed as a
+			  bus slave of the Shared Peripheral Bus and when two
+			  or more bus masters (CPU, DMA or DSP) try to access
+			  it. This property is optional depending on the SoC
+			  design.
 
    - big-endian		: If this property is absent, the little endian mode
 			  will be in use as default. Otherwise, the big endian

Documentation/devicetree/bindings/sound/fsl,esai.txt

@@ -27,6 +27,11 @@ Required properties:
 			  derive HCK, SCK and FS.
 	"fsys"		  The system clock derived from ahb clock used to
 			  derive HCK, SCK and FS.
+	"spba"		  The spba clock is required when ESAI is placed as a
+			  bus slave of the Shared Peripheral Bus and when two
+			  or more bus masters (CPU, DMA or DSP) try to access
+			  it. This property is optional depending on the SoC
+			  design.
 
   - fsl,fifo-depth	: The number of elements in the transmit and receive
 			  FIFOs. This number is the maximum allowed value for

Documentation/devicetree/bindings/sound/fsl,spdif.txt

@@ -27,6 +27,11 @@ Required properties:
 			  Transceiver Clock Diagram" of SoC reference manual.
 			  It can also be referred to TxClk_Source bit of
 			  register SPDIF_STC.
+	"spba"		  The spba clock is required when SPDIF is placed as a
+			  bus slave of the Shared Peripheral Bus and when two
+			  or more bus masters (CPU, DMA or DSP) try to access
+			  it. This property is optional depending on the SoC
+			  design.
 
    - big-endian		: If this property is absent, the native endian mode
 			  will be in use as default, or the big endian mode

Documentation/devicetree/bindings/sound/img,i2s-in.txt

+Imagination Technologies I2S Input Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,i2s-in"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Must include the following entry:
+	"sys"	The system clock
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"rx"	Single DMA channel used by all active I2S channels
+
+  - img,i2s-channels : Number of I2S channels instantiated in the I2S in block
+
+Optional Properties:
+
+  - interrupts : Contains the I2S in interrupts. Depending on
+	the configuration, there may be no interrupts, one interrupt,
+	or an interrupt per I2S channel. For the case where there is
+	one interrupt per channel, the interrupts should be listed
+	in ascending channel order
+
+  - resets: Contains a phandle to the I2S in reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Example:
+
+i2s_in: i2s-in@18100800 {
+	compatible = "img,i2s-in";
+	reg = <0x18100800 0x200>;
+	interrupts = <GIC_SHARED 7 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 30 0xffffffff 0>;
+	dma-names = "rx";
+	clocks = <&cr_periph SYS_CLK_I2S_IN>;
+	clock-names = "sys";
+	img,i2s-channels = <6>;
+	#sound-dai-cells = <0>;
+};

Documentation/devicetree/bindings/sound/img,i2s-out.txt

+Imagination Technologies I2S Output Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,i2s-out"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Must include the following entries:
+	"sys"	The system clock
+	"ref"	The reference clock
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"tx"	Single DMA channel used by all active I2S channels
+
+  - img,i2s-channels : Number of I2S channels instantiated in the I2S out block
+
+  - resets: Contains a phandle to the I2S out reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Optional Properties:
+
+  - interrupts : Contains the I2S out interrupts. Depending on
+	the configuration, there may be no interrupts, one interrupt,
+	or an interrupt per I2S channel. For the case where there is
+	one interrupt per channel, the interrupts should be listed
+	in ascending channel order
+
+Example:
+
+i2s_out: i2s-out@18100A00 {
+	compatible = "img,i2s-out";
+	reg = <0x18100A00 0x200>;
+	interrupts = <GIC_SHARED 13 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 23 0xffffffff 0>;
+	dma-names = "tx";
+	clocks = <&cr_periph SYS_CLK_I2S_OUT>,
+		 <&clk_core CLK_I2S>;
+	clock-names = "sys", "ref";
+	img,i2s-channels = <6>;
+	resets = <&pistachio_reset PISTACHIO_RESET_I2S_OUT>;
+	reset-names = "rst";
+	#sound-dai-cells = <0>;
+};

Documentation/devicetree/bindings/sound/img,parallel-out.txt

+Imagination Technologies Parallel Output Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,parallel-out".
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device.
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"tx"
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"sys"	The system clock
+	"ref"	The reference clock
+
+  - resets: Contains a phandle to the parallel out reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Optional Properties:
+
+  - interrupts : Contains the parallel out interrupt, if present
+
+Example:
+
+parallel_out: parallel-out@18100C00 {
+	compatible = "img,parallel-out";
+	reg = <0x18100C00 0x100>;
+	interrupts = <GIC_SHARED 19 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 16 0xffffffff 0>;
+	dma-names = "tx";
+	clocks = <&cr_periph SYS_CLK_PAUD_OUT>,
+		 <&clk_core CLK_AUDIO_DAC>;
+	clock-names = "sys", "ref";
+	resets = <&pistachio_reset PISTACHIO_RESET_PRL_OUT>;
+	reset-names = "rst";
+	#sound-dai-cells = <0>;
+};

Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt

+Pistachio internal DAC DT bindings
+
+Required properties:
+
+  - compatible: "img,pistachio-internal-dac"
+
+  - img,cr-top : Must contain a phandle to the top level control syscon
+		 node which contains the internal dac control registers
+
+  - VDD-supply : Digital power supply regulator (+1.8V or +3.3V)
+
+Examples:
+
+internal_dac: internal-dac {
+	compatible = "img,pistachio-internal-dac";
+	img,cr-top = <&cr_top>;
+	VDD-supply = <&supply3v3>;
+};

Documentation/devicetree/bindings/sound/img,spdif-in.txt

+Imagination Technologies SPDIF Input Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,spdif-in"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"rx"
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Includes the following entries:
+	"sys"	The system clock
+
+Optional Properties:
+
+  - resets: Should contain a phandle to the spdif in reset signal, if any
+
+  - reset-names: Should contain the reset signal name "rst", if a
+	reset phandle is given
+
+  - interrupts : Contains the spdif in interrupt, if present
+
+Example:
+
+spdif_in: spdif-in@18100E00 {
+	compatible = "img,spdif-in";
+	reg = <0x18100E00 0x100>;
+	interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 15 0xffffffff 0>;
+	dma-names = "rx";
+	clocks = <&cr_periph SYS_CLK_SPDIF_IN>;
+	clock-names = "sys";
+	#sound-dai-cells = <0>;
+};

Documentation/devicetree/bindings/sound/img,spdif-out.txt

+Imagination Technologies SPDIF Output Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,spdif-out"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"tx"
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"sys"	The system clock
+	"ref"	The reference clock
+
+  - resets: Contains a phandle to the spdif out reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Optional Properties:
+
+  - interrupts : Contains the parallel out interrupt, if present
+
+Example:
+
+spdif_out: spdif-out@18100D00 {
+	compatible = "img,spdif-out";
+	reg = <0x18100D00 0x100>;
+	interrupts = <GIC_SHARED 21 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 14 0xffffffff 0>;
+	dma-names = "tx";
+	clocks = <&cr_periph SYS_CLK_SPDIF_OUT>,
+		 <&clk_core CLK_SPDIF>;
+	clock-names = "sys", "ref";
+	resets = <&pistachio_reset PISTACHIO_RESET_SPDIF_OUT>;
+	reset-names = "rst";
+	#sound-dai-cells = <0>;
+};

Documentation/devicetree/bindings/sound/inno-rk3036.txt

+Inno audio codec for RK3036
+
+Inno audio codec is integrated inside RK3036 SoC.
+
+Required properties:
+- compatible : Should be "rockchip,rk3036-codec".
+- reg : The registers of codec.
+- clock-names : Should be "acodec_pclk".
+- clocks : The clock of codec.
+- rockchip,grf : The phandle of grf device node.
+
+Example:
+
+	acodec: acodec-ana@20030000 {
+		compatible = "rk3036-codec";
+		reg = <0x20030000 0x4000>;
+		rockchip,grf = <&grf>;
+		clock-names = "acodec_pclk";
+		clocks = <&cru ACLK_VCODEC>;
+	};

Documentation/devicetree/bindings/sound/renesas,rsnd.txt

@@ -7,8 +7,11 @@ Required properties:
 				  "renesas,rcar_sound-gen3" if generation3
 				  Examples with soctypes are:
 				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
+				    - "renesas,rcar_sound-r8a7779" (R-Car H1)
 				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
 				    - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
+				    - "renesas,rcar_sound-r8a7793" (R-Car M2-N)
+				    - "renesas,rcar_sound-r8a7794" (R-Car E2)
 				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
 - reg				: Should contain the register physical address.
 				  required register is
@@ -34,6 +37,8 @@ Required properties:
 				  see below for detail.
 - #sound-dai-cells		: it must be 0 if your system is using single DAI
 				  it must be 1 if your system is using multi  DAI
+
+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
@@ -244,3 +249,80 @@ rcar_sound: sound@ec500000 {
 		};
 	};
 };
+
+Example: simple sound card
+
+	rsnd_ak4643: sound {
+		compatible = "simple-audio-card";
+
+		simple-audio-card,format = "left_j";
+		simple-audio-card,bitclock-master = <&sndcodec>;
+		simple-audio-card,frame-master = <&sndcodec>;
+
+		sndcpu: simple-audio-card,cpu {
+			sound-dai = <&rcar_sound>;
+		};
+
+		sndcodec: simple-audio-card,codec {
+			sound-dai = <&ak4643>;
+			clocks = <&audio_clock>;
+		};
+	};
+
+&rcar_sound {
+	pinctrl-0 = <&sound_pins &sound_clk_pins>;
+	pinctrl-names = "default";
+
+	/* Single DAI */
+	#sound-dai-cells = <0>;
+
+	status = "okay";
+
+	rcar_sound,dai {
+		dai0 {
+			playback = <&ssi0 &src2 &dvc0>;
+			capture  = <&ssi1 &src3 &dvc1>;
+		};
+	};
+};
+
+&ssi1 {
+	shared-pin;
+};
+
+Example: simple sound card for TDM
+
+	rsnd_tdm: sound {
+		compatible = "simple-audio-card";
+
+		simple-audio-card,format = "left_j";
+		simple-audio-card,bitclock-master = <&sndcodec>;
+		simple-audio-card,frame-master = <&sndcodec>;
+
+		sndcpu: simple-audio-card,cpu {
+			sound-dai = <&rcar_sound>;
+			dai-tdm-slot-num = <6>;
+		};
+
+		sndcodec: simple-audio-card,codec {
+			sound-dai = <&xxx>;
+		};
+	};
+
+Example: simple sound card for Multi channel
+
+&rcar_sound {
+	pinctrl-0 = <&sound_pins &sound_clk_pins>;
+	pinctrl-names = "default";
+
+	/* Single DAI */
+	#sound-dai-cells = <0>;
+
+	status = "okay";
+
+	rcar_sound,dai {
+		dai0 {
+			playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>;
+		};
+	};
+};

Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt

@@ -4,8 +4,8 @@ Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC
 
 Required properties:
 
-- compatible				: "renesas,rsrc-card,<board>"
-					  Examples with soctypes are:
+- compatible				: "renesas,rsrc-card{,<board>}"
+					  Examples with boards are:
 					    - "renesas,rsrc-card"
 					    - "renesas,rsrc-card,lager"
 					    - "renesas,rsrc-card,koelsch"

Documentation/devicetree/bindings/sound/rockchip-i2s.txt

@@ -19,6 +19,7 @@ Required properties:
 - clock-names: should contain followings:
    - "i2s_hclk": clock for I2S BUS
    - "i2s_clk" : clock for I2S controller
+- rockchip,playback-channels: max playback channels, if not set, 8 channels default.
 - rockchip,capture-channels: max capture channels, if not set, 2 channels default.
 
 Example for rk3288 I2S controller:
@@ -31,5 +32,6 @@ i2s@ff890000 {
 	dma-names = "tx", "rx";
 	clock-names = "i2s_hclk", "i2s_clk";
 	clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
+	rockchip,playback-channels = <8>;
 	rockchip,capture-channels = <2>;
 };

Documentation/devicetree/bindings/sound/rt5616.txt

+RT5616 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5616".
+
+- reg : The I2C address of the device.
+
+Pins on the device (for linking into audio routes) for RT5616:
+
+  * IN1P
+  * IN2P
+  * IN2N
+  * LOUTL
+  * LOUTR
+  * HPOL
+  * HPOR
+
+Example:
+
+codec: rt5616@1b {
+	compatible = "realtek,rt5616";
+	reg = <0x1b>;
+};

Documentation/devicetree/bindings/sound/rt5659.txt

+RT5659/RT5658 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : One of "realtek,rt5659" or "realtek,rt5658".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- realtek,in1-differential
+- realtek,in3-differential
+- realtek,in4-differential
+  Boolean. Indicate MIC1/3/4 input are differential, rather than single-ended.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using IN2N pin as dmic1 data pin
+  2: using GPIO5 pin as dmic1 data pin
+  3: using GPIO9 pin as dmic1 data pin
+  4: using GPIO11 pin as dmic1 data pin
+
+- realtek,dmic2-data-pin
+  0: dmic2 is not used
+  1: using IN2P pin as dmic2 data pin
+  2: using GPIO6 pin as dmic2 data pin
+  3: using GPIO10 pin as dmic2 data pin
+  4: using GPIO12 pin as dmic2 data pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD3 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+- realtek,reset-gpios : The GPIO that controls the CODEC's RESET pin.
+
+Pins on the device (for linking into audio routes) for RT5659/RT5658:
+
+  * DMIC L1
+  * DMIC R1
+  * DMIC L2
+  * DMIC R2
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * IN3P
+  * IN3N
+  * IN4P
+  * IN4N
+  * HPOL
+  * HPOR
+  * SPOL
+  * SPOR
+  * LOUTL
+  * LOUTR
+  * MONOOUT
+  * PDML
+  * PDMR
+  * SPDIF
+
+Example:
+
+rt5659 {
+	compatible = "realtek,rt5659";
+	reg = <0x1b>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+};

Documentation/devicetree/bindings/sound/rt5677.txt

@@ -18,7 +18,7 @@ Required properties:
 Optional properties:
 
 - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin.
-- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin.
+- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low.
 
 - realtek,in1-differential
 - realtek,in2-differential

Documentation/devicetree/bindings/sound/sun4i-codec.txt

@@ -14,6 +14,9 @@ Required properties:
    - "apb": the parent APB clock for this controller
    - "codec": the parent module clock
 
+Optional properties:
+- allwinner,pa-gpios: gpio to enable external amplifier
+
 Example:
 codec: codec@01c22c00 {
 	#sound-dai-cells = <0>;

Documentation/devicetree/bindings/sound/ti,pcm3168a.txt

+Texas Instruments pcm3168a DT bindings
+
+This driver supports both SPI and I2C bus access for this codec
+
+Required properties:
+
+  - compatible: "ti,pcm3168a"
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Includes the following entries:
+	"scki"	The system clock
+
+  - VDD1-supply : Digital power supply regulator 1 (+3.3V)
+
+  - VDD2-supply : Digital power supply regulator 2 (+3.3V)
+
+  - VCCAD1-supply : ADC power supply regulator 1 (+5V)
+
+  - VCCAD2-supply : ADC power supply regulator 2 (+5V)
+
+  - VCCDA1-supply : DAC power supply regulator 1 (+5V)
+
+  - VCCDA2-supply : DAC power supply regulator 2 (+5V)
+
+For required properties on SPI/I2C, consult SPI/I2C device tree documentation
+
+Examples:
+
+i2c0: i2c0@0 {
+
+	...
+
+	pcm3168a: audio-codec@44 {
+		compatible = "ti,pcm3168a";
+		reg = <0x44>;
+		clocks = <&clk_core CLK_AUDIO>;
+		clock-names = "scki";
+		VDD1-supply = <&supply3v3>;
+		VDD2-supply = <&supply3v3>;
+		VCCAD1-supply = <&supply5v0>;
+		VCCAD2-supply = <&supply5v0>;
+		VCCDA1-supply = <&supply5v0>;
+		VCCDA2-supply = <&supply5v0>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&dac_clk_pin>;
+	};
+};

Documentation/devicetree/bindings/sound/wlf,wm8974.txt

+WM8974 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+  - compatible: "wlf,wm8974"
+  - reg: the I2C address or SPI chip select number of the device
+
+Examples:
+
+codec: wm8974@1a {
+	compatible = "wlf,wm8974";
+	reg = <0x1a>;
+};

Documentation/sound/alsa/img,spdif-in.txt

+The Imagination Technologies SPDIF Input controller contains the following
+controls:
+
+name='IEC958 Capture Mask',index=0
+
+This control returns a mask that shows which of the IEC958 status bits
+can be read using the 'IEC958 Capture Default' control.
+
+name='IEC958 Capture Default',index=0
+
+This control returns the status bits contained within the SPDIF stream that
+is being received. The 'IEC958 Capture Mask' shows which bits can be read
+from this control.
+
+name='SPDIF In Multi Frequency Acquire',index=0
+name='SPDIF In Multi Frequency Acquire',index=1
+name='SPDIF In Multi Frequency Acquire',index=2
+name='SPDIF In Multi Frequency Acquire',index=3
+
+This control is used to attempt acquisition of up to four different sample
+rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency'
+control.
+
+When the value of this control is set to {0,0,0,0}, the rate given to hw_params
+will determine the single rate the block will capture. Else, the rate given to
+hw_params will be ignored, and the block will attempt capture for each of the
+four sample rates set here.
+
+If less than four rates are required, the same rate can be specified more than
+once
+
+name='SPDIF In Lock Frequency',index=0
+
+This control returns the active capture rate, or 0 if a lock has not been
+acquired
+
+name='SPDIF In Lock TRK',index=0
+
+This control is used to modify the locking/jitter rejection characteristics
+of the block. Larger values increase the locking range, but reduce jitter
+rejection.
+
+name='SPDIF In Lock Acquire Threshold',index=0
+
+This control is used to change the threshold at which a lock is acquired.
+
+name='SPDIF In Lock Release Threshold',index=0
+
+This control is used to change the threshold at which a lock is released.

arch/arm/mach-s3c64xx/dev-audio.c

@@ -54,12 +54,13 @@ static int s3c64xx_i2s_cfg_gpio(struct platform_device *pdev)
 
 static struct resource s3c64xx_iis0_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C64XX_PA_IIS0, SZ_256),
-	[1] = DEFINE_RES_DMA(DMACH_I2S0_OUT),
-	[2] = DEFINE_RES_DMA(DMACH_I2S0_IN),
 };
 
-static struct s3c_audio_pdata i2sv3_pdata = {
+static struct s3c_audio_pdata i2s0_pdata = {
 	.cfg_gpio = s3c64xx_i2s_cfg_gpio,
+	.dma_filter = pl08x_filter_id,
+	.dma_playback = DMACH_I2S0_OUT,
+	.dma_capture = DMACH_I2S0_IN,
 };
 
 struct platform_device s3c64xx_device_iis0 = {
@@ -68,15 +69,20 @@ struct platform_device s3c64xx_device_iis0 = {
 	.num_resources	  = ARRAY_SIZE(s3c64xx_iis0_resource),
 	.resource	  = s3c64xx_iis0_resource,
 	.dev = {
-		.platform_data = &i2sv3_pdata,
+		.platform_data = &i2s0_pdata,
 	},
 };
 EXPORT_SYMBOL(s3c64xx_device_iis0);
 
 static struct resource s3c64xx_iis1_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C64XX_PA_IIS1, SZ_256),
-	[1] = DEFINE_RES_DMA(DMACH_I2S1_OUT),
-	[2] = DEFINE_RES_DMA(DMACH_I2S1_IN),
+};
+
+static struct s3c_audio_pdata i2s1_pdata = {
+	.cfg_gpio = s3c64xx_i2s_cfg_gpio,
+	.dma_filter = pl08x_filter_id,
+	.dma_playback = DMACH_I2S1_OUT,
+	.dma_capture = DMACH_I2S1_IN,
 };
 
 struct platform_device s3c64xx_device_iis1 = {
@@ -85,19 +91,20 @@ struct platform_device s3c64xx_device_iis1 = {
 	.num_resources	  = ARRAY_SIZE(s3c64xx_iis1_resource),
 	.resource	  = s3c64xx_iis1_resource,
 	.dev = {
-		.platform_data = &i2sv3_pdata,
+		.platform_data = &i2s1_pdata,
 	},
 };
 EXPORT_SYMBOL(s3c64xx_device_iis1);
 
 static struct resource s3c64xx_iisv4_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C64XX_PA_IISV4, SZ_256),
-	[1] = DEFINE_RES_DMA(DMACH_HSI_I2SV40_TX),
-	[2] = DEFINE_RES_DMA(DMACH_HSI_I2SV40_RX),
 };
 
 static struct s3c_audio_pdata i2sv4_pdata = {
 	.cfg_gpio = s3c64xx_i2s_cfg_gpio,
+	.dma_filter = pl08x_filter_id,
+	.dma_playback = DMACH_HSI_I2SV40_TX,
+	.dma_capture = DMACH_HSI_I2SV40_RX,
 	.type = {
 		.i2s = {
 			.quirks = QUIRK_PRI_6CHAN,
@@ -142,12 +149,13 @@ static int s3c64xx_pcm_cfg_gpio(struct platform_device *pdev)
 
 static struct resource s3c64xx_pcm0_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C64XX_PA_PCM0, SZ_256),
-	[1] = DEFINE_RES_DMA(DMACH_PCM0_TX),
-	[2] = DEFINE_RES_DMA(DMACH_PCM0_RX),
 };
 
 static struct s3c_audio_pdata s3c_pcm0_pdata = {
 	.cfg_gpio = s3c64xx_pcm_cfg_gpio,
+	.dma_filter = pl08x_filter_id,
+	.dma_capture = DMACH_PCM0_RX,
+	.dma_playback = DMACH_PCM0_TX,
 };
 
 struct platform_device s3c64xx_device_pcm0 = {
@@ -163,12 +171,13 @@ EXPORT_SYMBOL(s3c64xx_device_pcm0);
 
 static struct resource s3c64xx_pcm1_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C64XX_PA_PCM1, SZ_256),
-	[1] = DEFINE_RES_DMA(DMACH_PCM1_TX),
-	[2] = DEFINE_RES_DMA(DMACH_PCM1_RX),
 };
 
 static struct s3c_audio_pdata s3c_pcm1_pdata = {
 	.cfg_gpio = s3c64xx_pcm_cfg_gpio,
+	.dma_filter = pl08x_filter_id,
+	.dma_playback = DMACH_PCM1_TX,
+	.dma_capture = DMACH_PCM1_RX,
 };
 
 struct platform_device s3c64xx_device_pcm1 = {
@@ -196,13 +205,15 @@ static int s3c64xx_ac97_cfg_gpe(struct platform_device *pdev)
 
 static struct resource s3c64xx_ac97_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C64XX_PA_AC97, SZ_256),
-	[1] = DEFINE_RES_DMA(DMACH_AC97_PCMOUT),
-	[2] = DEFINE_RES_DMA(DMACH_AC97_PCMIN),
-	[3] = DEFINE_RES_DMA(DMACH_AC97_MICIN),
-	[4] = DEFINE_RES_IRQ(IRQ_AC97),
+	[1] = DEFINE_RES_IRQ(IRQ_AC97),
 };
 
-static struct s3c_audio_pdata s3c_ac97_pdata;
+static struct s3c_audio_pdata s3c_ac97_pdata = {
+	.dma_playback = DMACH_AC97_PCMOUT,
+	.dma_filter = pl08x_filter_id,
+	.dma_capture = DMACH_AC97_PCMIN,
+	.dma_capture_mic = DMACH_AC97_MICIN,
+};
 
 static u64 s3c64xx_ac97_dmamask = DMA_BIT_MASK(32);
 

arch/arm/mach-s3c64xx/include/mach/dma.h

@@ -14,38 +14,38 @@
 #define S3C64XX_DMA_CHAN(name)		((unsigned long)(name))
 
 /* DMA0/SDMA0 */
-#define DMACH_UART0		S3C64XX_DMA_CHAN("uart0_tx")
-#define DMACH_UART0_SRC2	S3C64XX_DMA_CHAN("uart0_rx")
-#define DMACH_UART1		S3C64XX_DMA_CHAN("uart1_tx")
-#define DMACH_UART1_SRC2	S3C64XX_DMA_CHAN("uart1_rx")
-#define DMACH_UART2		S3C64XX_DMA_CHAN("uart2_tx")
-#define DMACH_UART2_SRC2	S3C64XX_DMA_CHAN("uart2_rx")
-#define DMACH_UART3		S3C64XX_DMA_CHAN("uart3_tx")
-#define DMACH_UART3_SRC2	S3C64XX_DMA_CHAN("uart3_rx")
-#define DMACH_PCM0_TX		S3C64XX_DMA_CHAN("pcm0_tx")
-#define DMACH_PCM0_RX		S3C64XX_DMA_CHAN("pcm0_rx")
-#define DMACH_I2S0_OUT		S3C64XX_DMA_CHAN("i2s0_tx")
-#define DMACH_I2S0_IN		S3C64XX_DMA_CHAN("i2s0_rx")
+#define DMACH_UART0		"uart0_tx"
+#define DMACH_UART0_SRC2	"uart0_rx"
+#define DMACH_UART1		"uart1_tx"
+#define DMACH_UART1_SRC2	"uart1_rx"
+#define DMACH_UART2		"uart2_tx"
+#define DMACH_UART2_SRC2	"uart2_rx"
+#define DMACH_UART3		"uart3_tx"
+#define DMACH_UART3_SRC2	"uart3_rx"
+#define DMACH_PCM0_TX		"pcm0_tx"
+#define DMACH_PCM0_RX		"pcm0_rx"
+#define DMACH_I2S0_OUT		"i2s0_tx"
+#define DMACH_I2S0_IN		"i2s0_rx"
 #define DMACH_SPI0_TX		S3C64XX_DMA_CHAN("spi0_tx")
 #define DMACH_SPI0_RX		S3C64XX_DMA_CHAN("spi0_rx")
-#define DMACH_HSI_I2SV40_TX	S3C64XX_DMA_CHAN("i2s2_tx")
-#define DMACH_HSI_I2SV40_RX	S3C64XX_DMA_CHAN("i2s2_rx")
+#define DMACH_HSI_I2SV40_TX	"i2s2_tx"
+#define DMACH_HSI_I2SV40_RX	"i2s2_rx"
 
 /* DMA1/SDMA1 */
-#define DMACH_PCM1_TX		S3C64XX_DMA_CHAN("pcm1_tx")
-#define DMACH_PCM1_RX		S3C64XX_DMA_CHAN("pcm1_rx")
-#define DMACH_I2S1_OUT		S3C64XX_DMA_CHAN("i2s1_tx")
-#define DMACH_I2S1_IN		S3C64XX_DMA_CHAN("i2s1_rx")
+#define DMACH_PCM1_TX		"pcm1_tx"
+#define DMACH_PCM1_RX		"pcm1_rx"
+#define DMACH_I2S1_OUT		"i2s1_tx"
+#define DMACH_I2S1_IN		"i2s1_rx"
 #define DMACH_SPI1_TX		S3C64XX_DMA_CHAN("spi1_tx")
 #define DMACH_SPI1_RX		S3C64XX_DMA_CHAN("spi1_rx")
-#define DMACH_AC97_PCMOUT	S3C64XX_DMA_CHAN("ac97_out")
-#define DMACH_AC97_PCMIN	S3C64XX_DMA_CHAN("ac97_in")
-#define DMACH_AC97_MICIN	S3C64XX_DMA_CHAN("ac97_mic")
-#define DMACH_PWM		S3C64XX_DMA_CHAN("pwm")
-#define DMACH_IRDA		S3C64XX_DMA_CHAN("irda")
-#define DMACH_EXTERNAL		S3C64XX_DMA_CHAN("external")
-#define DMACH_SECURITY_RX	S3C64XX_DMA_CHAN("sec_rx")
-#define DMACH_SECURITY_TX	S3C64XX_DMA_CHAN("sec_tx")
+#define DMACH_AC97_PCMOUT	"ac97_out"
+#define DMACH_AC97_PCMIN	"ac97_in"
+#define DMACH_AC97_MICIN	"ac97_mic"
+#define DMACH_PWM		"pwm"
+#define DMACH_IRDA		"irda"
+#define DMACH_EXTERNAL		"external"
+#define DMACH_SECURITY_RX	"sec_rx"
+#define DMACH_SECURITY_TX	"sec_tx"
 
 enum dma_ch {
 	DMACH_MAX = 32

arch/arm/plat-samsung/devs.c

@@ -65,6 +65,7 @@
 #include <linux/platform_data/usb-ohci-s3c2410.h>
 #include <plat/usb-phy.h>
 #include <plat/regs-spi.h>
+#include <linux/platform_data/asoc-s3c.h>
 #include <linux/platform_data/spi-s3c64xx.h>
 
 static u64 samsung_device_dma_mask = DMA_BIT_MASK(32);
@@ -74,9 +75,15 @@ static u64 samsung_device_dma_mask = DMA_BIT_MASK(32);
 static struct resource s3c_ac97_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C2440_PA_AC97, S3C2440_SZ_AC97),
 	[1] = DEFINE_RES_IRQ(IRQ_S3C244X_AC97),
-	[2] = DEFINE_RES_DMA_NAMED(DMACH_PCM_OUT, "PCM out"),
-	[3] = DEFINE_RES_DMA_NAMED(DMACH_PCM_IN, "PCM in"),
-	[4] = DEFINE_RES_DMA_NAMED(DMACH_MIC_IN, "Mic in"),
+};
+
+static struct s3c_audio_pdata s3c_ac97_pdata = {
+#ifdef CONFIG_S3C24XX_DMAC
+	.dma_filter = s3c24xx_dma_filter,
+#endif
+	.dma_playback = (void *)DMACH_PCM_OUT,
+	.dma_capture = (void *)DMACH_PCM_IN,
+	.dma_capture_mic = (void *)DMACH_MIC_IN,
 };
 
 struct platform_device s3c_device_ac97 = {
@@ -87,6 +94,7 @@ struct platform_device s3c_device_ac97 = {
 	.dev		= {
 		.dma_mask		= &samsung_device_dma_mask,
 		.coherent_dma_mask	= DMA_BIT_MASK(32),
+		.platform_data		= &s3c_ac97_pdata,
 	}
 };
 #endif /* CONFIG_CPU_S3C2440 */
@@ -566,6 +574,14 @@ static struct resource s3c_iis_resource[] = {
 	[0] = DEFINE_RES_MEM(S3C24XX_PA_IIS, S3C24XX_SZ_IIS),
 };
 
+static struct s3c_audio_pdata s3c_iis_platdata = {
+#ifdef CONFIG_S3C24XX_DMAC
+	.dma_filter = s3c24xx_dma_filter,
+#endif
+	.dma_playback = (void *)DMACH_I2S_OUT,
+	.dma_capture = (void *)DMACH_I2S_IN,
+};
+
 struct platform_device s3c_device_iis = {
 	.name		= "s3c24xx-iis",
 	.id		= -1,
@@ -574,6 +590,7 @@ struct platform_device s3c_device_iis = {
 	.dev		= {
 		.dma_mask		= &samsung_device_dma_mask,
 		.coherent_dma_mask	= DMA_BIT_MASK(32),
+		.platform_data		= &s3c_iis_platdata,
 	}
 };
 #endif /* CONFIG_PLAT_S3C24XX */

arch/x86/include/asm/platform_sst_audio.h

@@ -55,6 +55,7 @@ enum sst_audio_device_id_mrfld {
 	PIPE_MEDIA0_IN = 0x8F,
 	PIPE_MEDIA1_IN = 0x90,
 	PIPE_MEDIA2_IN = 0x91,
+	PIPE_MEDIA3_IN = 0x9C,
 	PIPE_RSVD = 0xFF,
 };
 

drivers/dma/Kconfig

@@ -432,7 +432,7 @@ config STE_DMA40
 	  Support for ST-Ericsson DMA40 controller
 
 config S3C24XX_DMAC
-	tristate "Samsung S3C24XX DMA support"
+	bool "Samsung S3C24XX DMA support"
 	depends on ARCH_S3C24XX
 	select DMA_ENGINE
 	select DMA_VIRTUAL_CHANNELS

include/linux/platform_data/asoc-s3c.h

@@ -13,6 +13,9 @@
  */
 #define S3C64XX_AC97_GPD  0
 #define S3C64XX_AC97_GPE  1
+
+#include <linux/dmaengine.h>
+
 extern void s3c64xx_ac97_setup_gpio(int);
 
 struct samsung_i2s {
@@ -39,6 +42,11 @@ struct samsung_i2s {
  */
 struct s3c_audio_pdata {
 	int (*cfg_gpio)(struct platform_device *);
+	dma_filter_fn dma_filter;
+	void *dma_playback;
+	void *dma_capture;
+	void *dma_play_sec;
+	void *dma_capture_mic;
 	union {
 		struct samsung_i2s i2s;
 	} type;

include/sound/ac97_codec.h

@@ -417,11 +417,13 @@
 #define AC97_RATES_MIC_ADC	4
 #define AC97_RATES_SPDIF	5
 
+#define AC97_NUM_GPIOS		16
 /*
  *
  */
 
 struct snd_ac97;
+struct snd_ac97_gpio_priv;
 struct snd_pcm_chmap;
 
 struct snd_ac97_build_ops {
@@ -529,6 +531,7 @@ struct snd_ac97 {
 	struct delayed_work power_work;
 #endif
 	struct device dev;
+	struct snd_ac97_gpio_priv *gpio_priv;
 
 	struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */
 };

include/sound/da7218.h

+/*
+ * da7218.h - DA7218 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#ifndef _DA7218_PDATA_H
+#define _DA7218_PDATA_H
+
+/* Mic Bias */
+enum da7218_micbias_voltage {
+	DA7218_MICBIAS_1_2V = -1,
+	DA7218_MICBIAS_1_6V,
+	DA7218_MICBIAS_1_8V,
+	DA7218_MICBIAS_2_0V,
+	DA7218_MICBIAS_2_2V,
+	DA7218_MICBIAS_2_4V,
+	DA7218_MICBIAS_2_6V,
+	DA7218_MICBIAS_2_8V,
+	DA7218_MICBIAS_3_0V,
+};
+
+enum da7218_mic_amp_in_sel {
+	DA7218_MIC_AMP_IN_SEL_DIFF = 0,
+	DA7218_MIC_AMP_IN_SEL_SE_P,
+	DA7218_MIC_AMP_IN_SEL_SE_N,
+};
+
+/* DMIC */
+enum da7218_dmic_data_sel {
+	DA7218_DMIC_DATA_LRISE_RFALL = 0,
+	DA7218_DMIC_DATA_LFALL_RRISE,
+};
+
+enum da7218_dmic_samplephase {
+	DA7218_DMIC_SAMPLE_ON_CLKEDGE = 0,
+	DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE,
+};
+
+enum da7218_dmic_clk_rate {
+	DA7218_DMIC_CLK_3_0MHZ = 0,
+	DA7218_DMIC_CLK_1_5MHZ,
+};
+
+/* Headphone Detect */
+enum da7218_hpldet_jack_rate {
+	DA7218_HPLDET_JACK_RATE_5US = 0,
+	DA7218_HPLDET_JACK_RATE_10US,
+	DA7218_HPLDET_JACK_RATE_20US,
+	DA7218_HPLDET_JACK_RATE_40US,
+	DA7218_HPLDET_JACK_RATE_80US,
+	DA7218_HPLDET_JACK_RATE_160US,
+	DA7218_HPLDET_JACK_RATE_320US,
+	DA7218_HPLDET_JACK_RATE_640US,
+};
+
+enum da7218_hpldet_jack_debounce {
+	DA7218_HPLDET_JACK_DEBOUNCE_OFF = 0,
+	DA7218_HPLDET_JACK_DEBOUNCE_2,
+	DA7218_HPLDET_JACK_DEBOUNCE_3,
+	DA7218_HPLDET_JACK_DEBOUNCE_4,
+};
+
+enum da7218_hpldet_jack_thr {
+	DA7218_HPLDET_JACK_THR_84PCT = 0,
+	DA7218_HPLDET_JACK_THR_88PCT,
+	DA7218_HPLDET_JACK_THR_92PCT,
+	DA7218_HPLDET_JACK_THR_96PCT,
+};
+
+struct da7218_hpldet_pdata {
+	enum da7218_hpldet_jack_rate jack_rate;
+	enum da7218_hpldet_jack_debounce jack_debounce;
+	enum da7218_hpldet_jack_thr jack_thr;
+	bool comp_inv;
+	bool hyst;
+	bool discharge;
+};
+
+struct da7218_pdata {
+	/* Mic */
+	enum da7218_micbias_voltage micbias1_lvl;
+	enum da7218_micbias_voltage micbias2_lvl;
+	enum da7218_mic_amp_in_sel mic1_amp_in_sel;
+	enum da7218_mic_amp_in_sel mic2_amp_in_sel;
+
+	/* DMIC */
+	enum da7218_dmic_data_sel dmic1_data_sel;
+	enum da7218_dmic_data_sel dmic2_data_sel;
+	enum da7218_dmic_samplephase dmic1_samplephase;
+	enum da7218_dmic_samplephase dmic2_samplephase;
+	enum da7218_dmic_clk_rate dmic1_clk_rate;
+	enum da7218_dmic_clk_rate dmic2_clk_rate;
+
+	/* HP Diff Supply - DA7217 only */
+	bool hp_diff_single_supply;
+
+	/* HP Detect - DA7218 only */
+	struct da7218_hpldet_pdata *hpldet_pdata;
+};
+
+#endif /* _DA7218_PDATA_H */

include/sound/da7219.h

@@ -14,17 +14,10 @@
 #ifndef __DA7219_PDATA_H
 #define __DA7219_PDATA_H
 
-/* LDO */
-enum da7219_ldo_lvl_sel {
-	DA7219_LDO_LVL_SEL_1_05V = 0,
-	DA7219_LDO_LVL_SEL_1_10V,
-	DA7219_LDO_LVL_SEL_1_20V,
-	DA7219_LDO_LVL_SEL_1_40V,
-};
-
 /* Mic Bias */
 enum da7219_micbias_voltage {
-	DA7219_MICBIAS_1_8V = 1,
+	DA7219_MICBIAS_1_6V = 0,
+	DA7219_MICBIAS_1_8V,
 	DA7219_MICBIAS_2_0V,
 	DA7219_MICBIAS_2_2V,
 	DA7219_MICBIAS_2_4V,
@@ -41,9 +34,6 @@ enum da7219_mic_amp_in_sel {
 struct da7219_aad_pdata;
 
 struct da7219_pdata {
-	/* Internal LDO */
-	enum da7219_ldo_lvl_sel ldo_lvl_sel;
-
 	/* Mic */
 	enum da7219_micbias_voltage micbias_lvl;
 	enum da7219_mic_amp_in_sel mic_amp_in_sel;

include/sound/designware_i2s.h

@@ -45,6 +45,11 @@ struct i2s_platform_data {
 	u32 snd_fmts;
 	u32 snd_rates;
 
+	#define DW_I2S_QUIRK_COMP_REG_OFFSET	(1 << 0)
+	unsigned int quirks;
+	unsigned int i2s_reg_comp1;
+	unsigned int i2s_reg_comp2;
+
 	void *play_dma_data;
 	void *capture_dma_data;
 	bool (*filter)(struct dma_chan *chan, void *slave);

include/sound/hdaudio_ext.h

@@ -186,9 +186,15 @@ struct hdac_ext_device {
 	/* codec ops */
 	struct hdac_ext_codec_ops ops;
 
+	struct snd_card *card;
+	void *scodec;
 	void *private_data;
 };
 
+struct hdac_ext_dma_params {
+	u32 format;
+	u8 stream_tag;
+};
 #define to_ehdac_device(dev) (container_of((dev), \
 				 struct hdac_ext_device, hdac))
 /*

include/sound/rt5659.h

+/*
+ * linux/sound/rt5659.h -- Platform data for RT5659
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5659_H
+#define __LINUX_SND_RT5659_H
+
+enum rt5659_dmic1_data_pin {
+	RT5659_DMIC1_NULL,
+	RT5659_DMIC1_DATA_IN2N,
+	RT5659_DMIC1_DATA_GPIO5,
+	RT5659_DMIC1_DATA_GPIO9,
+	RT5659_DMIC1_DATA_GPIO11,
+};
+
+enum rt5659_dmic2_data_pin {
+	RT5659_DMIC2_NULL,
+	RT5659_DMIC2_DATA_IN2P,
+	RT5659_DMIC2_DATA_GPIO6,
+	RT5659_DMIC2_DATA_GPIO10,
+	RT5659_DMIC2_DATA_GPIO12,
+};
+
+enum rt5659_jd_src {
+	RT5659_JD_NULL,
+	RT5659_JD3,
+};
+
+struct rt5659_platform_data {
+	bool in1_diff;
+	bool in3_diff;
+	bool in4_diff;
+
+	int ldo1_en; /* GPIO for LDO1_EN */
+	int reset; /* GPIO for RESET */
+
+	enum rt5659_dmic1_data_pin dmic1_data_pin;
+	enum rt5659_dmic2_data_pin dmic2_data_pin;
+	enum rt5659_jd_src jd_src;
+};
+
+#endif
+

include/sound/soc-dapm.h

@@ -49,6 +49,9 @@ struct device;
 #define SND_SOC_DAPM_SIGGEN(wname) \
 {	.id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
+#define SND_SOC_DAPM_SINK(wname) \
+{	.id = snd_soc_dapm_sink, .name = wname, .kcontrol_news = NULL, \
+	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
 #define SND_SOC_DAPM_INPUT(wname) \
 {	.id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
@@ -485,6 +488,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_aif_in,		/* audio interface input */
 	snd_soc_dapm_aif_out,		/* audio interface output */
 	snd_soc_dapm_siggen,		/* signal generator */
+	snd_soc_dapm_sink,
 	snd_soc_dapm_dai_in,		/* link to DAI structure */
 	snd_soc_dapm_dai_out,
 	snd_soc_dapm_dai_link,		/* link between two DAI structures */

include/sound/soc-topology.h

@@ -92,8 +92,10 @@ struct snd_soc_tplg_kcontrol_ops {
 /* Bytes ext operations, for TLV byte controls */
 struct snd_soc_tplg_bytes_ext_ops {
 	u32 id;
-	int (*get)(unsigned int __user *bytes, unsigned int size);
-	int (*put)(const unsigned int __user *bytes, unsigned int size);
+	int (*get)(struct snd_kcontrol *kcontrol, unsigned int __user *bytes,
+							unsigned int size);
+	int (*put)(struct snd_kcontrol *kcontrol,
+			const unsigned int __user *bytes, unsigned int size);
 };
 
 /*

include/sound/soc.h

@@ -110,6 +110,14 @@
 	.put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
 					  max, invert, 0) }
+#define SOC_DOUBLE_STS(xname, reg, shift_left, shift_right, max, invert) \
+{									\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),		\
+	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,		\
+	.access = SNDRV_CTL_ELEM_ACCESS_READ |				\
+		SNDRV_CTL_ELEM_ACCESS_VOLATILE,				\
+	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right,	\
+					  max, invert, 0) }
 #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.info = snd_soc_info_volsw, \
@@ -293,6 +301,9 @@
 		{.base = xbase, .num_regs = xregs,	      \
 		 .mask = xmask }) }
 
+/*
+ * SND_SOC_BYTES_EXT is deprecated, please USE SND_SOC_BYTES_TLV instead
+ */
 #define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_bytes_info_ext, \
@@ -1037,6 +1048,9 @@ struct snd_soc_dai_link {
 
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
+
+	struct list_head list; /* DAI link list of the soc card */
+	struct snd_soc_dobj dobj; /* For topology */
 };
 
 struct snd_soc_codec_conf {
@@ -1101,12 +1115,20 @@ struct snd_soc_card {
 				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
+	int (*add_dai_link)(struct snd_soc_card *,
+			    struct snd_soc_dai_link *link);
+	void (*remove_dai_link)(struct snd_soc_card *,
+			    struct snd_soc_dai_link *link);
+
 	long pmdown_time;
 
 	/* CPU <--> Codec DAI links  */
-	struct snd_soc_dai_link *dai_link;
-	int num_links;
-	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai_link *dai_link;  /* predefined links only */
+	int num_links;  /* predefined links only */
+	struct list_head dai_link_list; /* all links */
+	int num_dai_links;
+
+	struct list_head rtd_list;
 	int num_rtd;
 
 	/* optional codec specific configuration */
@@ -1201,6 +1223,9 @@ struct snd_soc_pcm_runtime {
 	struct dentry *debugfs_dpcm_root;
 	struct dentry *debugfs_dpcm_state;
 #endif
+
+	unsigned int num; /* 0-based and monotonic increasing */
+	struct list_head list; /* rtd list of the soc card */
 };
 
 /* mixer control */
@@ -1225,8 +1250,10 @@ struct soc_bytes_ext {
 	struct snd_soc_dobj dobj;
 
 	/* used for TLV byte control */
-	int (*get)(unsigned int __user *bytes, unsigned int size);
-	int (*put)(const unsigned int __user *bytes, unsigned int size);
+	int (*get)(struct snd_kcontrol *kcontrol, unsigned int __user *bytes,
+			unsigned int size);
+	int (*put)(struct snd_kcontrol *kcontrol, const unsigned int __user *bytes,
+			unsigned int size);
 };
 
 /* multi register control */
@@ -1644,6 +1671,11 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev,
 				   struct device_node *of_node,
 				   struct snd_soc_dai_link *dai_link);
 
+int snd_soc_add_dai_link(struct snd_soc_card *card,
+				struct snd_soc_dai_link *dai_link);
+void snd_soc_remove_dai_link(struct snd_soc_card *card,
+			     struct snd_soc_dai_link *dai_link);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS

include/uapi/sound/asoc.h

@@ -243,7 +243,7 @@ struct snd_soc_tplg_manifest {
 	__le32 control_elems;	/* number of control elements */
 	__le32 widget_elems;	/* number of widget elements */
 	__le32 graph_elems;	/* number of graph elements */
-	__le32 dai_elems;	/* number of DAI elements */
+	__le32 pcm_elems;	/* number of PCM elements */
 	__le32 dai_link_elems;	/* number of DAI link elements */
 	struct snd_soc_tplg_private priv;
 } __attribute__((packed));

include/uapi/sound/compress_params.h

@@ -73,7 +73,8 @@
 #define SND_AUDIOCODEC_IEC61937              ((__u32) 0x0000000B)
 #define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
 #define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
-#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_G729
+#define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_BESPOKE
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
@@ -312,7 +313,7 @@ struct snd_enc_flac {
 
 struct snd_enc_generic {
 	__u32 bw;	/* encoder bandwidth */
-	__s32 reserved[15];
+	__s32 reserved[15];	/* Can be used for SND_AUDIOCODEC_BESPOKE */
 } __attribute__((packed, aligned(4)));
 
 union snd_codec_options {

sound/soc/Kconfig

@@ -50,6 +50,7 @@ source "sound/soc/jz4740/Kconfig"
 source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
+source "sound/soc/img/Kconfig"
 source "sound/soc/intel/Kconfig"
 source "sound/soc/mediatek/Kconfig"
 source "sound/soc/mxs/Kconfig"

sound/soc/Makefile

@@ -27,6 +27,7 @@ obj-$(CONFIG_SND_SOC)	+= davinci/
 obj-$(CONFIG_SND_SOC)	+= dwc/
 obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
+obj-$(CONFIG_SND_SOC)	+= img/
 obj-$(CONFIG_SND_SOC)	+= intel/
 obj-$(CONFIG_SND_SOC)	+= mediatek/
 obj-$(CONFIG_SND_SOC)	+= mxs/

sound/soc/atmel/Kconfig

@@ -68,4 +68,13 @@ config SND_ATMEL_SOC_CLASSD
 	help
 	  Say Y if you want to add support for Atmel ASoC driver for boards using
 	  CLASSD.
+
+config SND_ATMEL_SOC_PDMIC
+	tristate "Atmel ASoC driver for boards using PDMIC"
+	depends on OF && (ARCH_AT91 || COMPILE_TEST)
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  Say Y if you want to add support for Atmel ASoC driver for boards using
+	  PDMIC.
 endif

sound/soc/atmel/Makefile

@@ -12,8 +12,10 @@ snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
 snd-atmel-soc-wm8904-objs := atmel_wm8904.o
 snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 snd-atmel-soc-classd-objs := atmel-classd.o
+snd-atmel-soc-pdmic-objs := atmel-pdmic.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
 obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
+obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o

sound/soc/atmel/atmel-classd.c

@@ -106,7 +106,7 @@ static const struct snd_pcm_hardware atmel_classd_hw = {
 	.rates			= ATMEL_CLASSD_RATES,
 	.rate_min		= 8000,
 	.rate_max		= 96000,
-	.channels_min		= 2,
+	.channels_min		= 1,
 	.channels_max		= 2,
 	.buffer_bytes_max	= 64 * 1024,
 	.period_bytes_min	= 256,
@@ -145,7 +145,7 @@ static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
 
 static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
 	.playback = {
-		.channels_min	= 2,
+		.channels_min	= 1,
 		.channels_max	= 2,
 		.rates		= ATMEL_CLASSD_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S16_LE,},
@@ -171,9 +171,13 @@ atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	if (params_channels(params) == 1)
+		slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	else
+		slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+
 	slave_config->direction		= DMA_MEM_TO_DEV;
 	slave_config->dst_addr		= dd->phy_base + CLASSD_THR;
-	slave_config->dst_addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES;
 	slave_config->dst_maxburst	= 1;
 	slave_config->src_maxburst	= 1;
 	slave_config->device_fc		= false;
@@ -486,7 +490,7 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = {
 	.name = ATMEL_CLASSD_CODEC_DAI_NAME,
 	.playback = {
 		.stream_name	= "Playback",
-		.channels_min	= 2,
+		.channels_min	= 1,
 		.channels_max	= 2,
 		.rates		= ATMEL_CLASSD_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S16_LE,
@@ -636,8 +640,10 @@ static int atmel_classd_probe(struct platform_device *pdev)
 
 	/* register sound card */
 	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
-	if (!card)
-		return -ENOMEM;
+	if (!card) {
+		ret = -ENOMEM;
+		goto unregister_codec;
+	}
 
 	snd_soc_card_set_drvdata(card, dd);
 	platform_set_drvdata(pdev, card);
@@ -645,16 +651,20 @@ static int atmel_classd_probe(struct platform_device *pdev)
 	ret = atmel_classd_asoc_card_init(dev, card);
 	if (ret) {
 		dev_err(dev, "failed to init sound card\n");
-		return ret;
+		goto unregister_codec;
 	}
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret) {
 		dev_err(dev, "failed to register sound card: %d\n", ret);
-		return ret;
+		goto unregister_codec;
 	}
 
 	return 0;
+
+unregister_codec:
+	snd_soc_unregister_codec(dev);
+	return ret;
 }
 
 static int atmel_classd_remove(struct platform_device *pdev)

sound/soc/atmel/atmel-pdmic.h

+#ifndef __ATMEL_PDMIC_H_
+#define __ATMEL_PDMIC_H_
+
+#include <linux/bitops.h>
+
+#define PDMIC_CR	0x00000000
+
+#define PDMIC_CR_SWRST		0x1
+#define PDMIC_CR_SWRST_MASK	BIT(0)
+#define PDMIC_CR_SWRST_SHIFT	(0)
+
+#define PDMIC_CR_ENPDM_DIS	0x0
+#define PDMIC_CR_ENPDM_EN	0x1
+#define PDMIC_CR_ENPDM_MASK	BIT(4)
+#define PDMIC_CR_ENPDM_SHIFT	(4)
+
+#define PDMIC_MR	0x00000004
+
+#define PDMIC_MR_CLKS_PCK	0x0
+#define PDMIC_MR_CLKS_GCK	0x1
+#define PDMIC_MR_CLKS_MASK	BIT(4)
+#define PDMIC_MR_CLKS_SHIFT	(4)
+
+#define PDMIC_MR_PRESCAL_MASK	GENMASK(14, 8)
+#define PDMIC_MR_PRESCAL_SHIFT	(8)
+
+#define PDMIC_CDR	0x00000014
+
+#define PDMIC_IER	0x00000018
+#define PDMIC_IER_OVRE			BIT(25)
+
+#define PDMIC_IDR	0x0000001c
+#define PDMIC_IDR_OVRE			BIT(25)
+
+#define PDMIC_IMR	0x00000020
+
+#define PDMIC_ISR	0x00000024
+#define PDMIC_ISR_OVRE			BIT(25)
+
+#define PDMIC_DSPR0	0x00000058
+
+#define PDMIC_DSPR0_HPFBYP_DIS		0x1
+#define PDMIC_DSPR0_HPFBYP_EN		0x0
+#define PDMIC_DSPR0_HPFBYP_MASK		BIT(1)
+#define PDMIC_DSPR0_HPFBYP_SHIFT	(1)
+
+#define PDMIC_DSPR0_SINBYP_DIS		0x1
+#define PDMIC_DSPR0_SINBYP_EN		0x0
+#define PDMIC_DSPR0_SINBYP_MASK		BIT(2)
+#define PDMIC_DSPR0_SINBYP_SHIFT	(2)
+
+#define PDMIC_DSPR0_SIZE_16_BITS	0x0
+#define PDMIC_DSPR0_SIZE_32_BITS	0x1
+#define PDMIC_DSPR0_SIZE_MASK		BIT(3)
+#define PDMIC_DSPR0_SIZE_SHIFT		(3)
+
+#define PDMIC_DSPR0_OSR_128		0x0
+#define PDMIC_DSPR0_OSR_64		0x1
+#define PDMIC_DSPR0_OSR_MASK		GENMASK(6, 4)
+#define PDMIC_DSPR0_OSR_SHIFT		(4)
+
+#define PDMIC_DSPR0_SCALE_MASK		GENMASK(11, 8)
+#define PDMIC_DSPR0_SCALE_SHIFT		(8)
+
+#define PDMIC_DSPR0_SHIFT_MASK		GENMASK(15, 12)
+#define PDMIC_DSPR0_SHIFT_SHIFT		(12)
+
+#define PDMIC_DSPR1	0x0000005c
+
+#define PDMIC_DSPR1_DGAIN_MASK		GENMASK(14, 0)
+#define PDMIC_DSPR1_DGAIN_SHIFT		(0)
+
+#define PDMIC_DSPR1_OFFSET_MASK		GENMASK(31, 16)
+#define PDMIC_DSPR1_OFFSET_SHIFT	(16)
+
+#define PDMIC_WPMR	0x000000e4
+
+#define PDMIC_WPSR	0x000000e8
+
+#endif

sound/soc/atmel/atmel_wm8904.c

@@ -183,6 +183,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = {
 	.driver = {
 		.name = "atmel-wm8904-audio",
 		.of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids),
+		.pm		= &snd_soc_pm_ops,
 	},
 	.probe = atmel_asoc_wm8904_probe,
 	.remove = atmel_asoc_wm8904_remove,

sound/soc/codecs/Kconfig

@@ -55,9 +55,11 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS4271_SPI if SPI_MASTER
 	select SND_SOC_CS42XX8_I2C if I2C
 	select SND_SOC_CS4349 if I2C
+	select SND_SOC_CS47L24 if MFD_CS47L24
 	select SND_SOC_CX20442 if TTY
 	select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_DA7213 if I2C
+	select SND_SOC_DA7218 if I2C
 	select SND_SOC_DA7219 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
@@ -66,7 +68,9 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ES8328_SPI if SPI_MASTER
 	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_GTM601
+	select SND_SOC_HDAC_HDMI
 	select SND_SOC_ICS43432
+	select SND_SOC_INNO_RK3036
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
@@ -85,14 +89,18 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM1681 if I2C
 	select SND_SOC_PCM1792A if SPI_MASTER
 	select SND_SOC_PCM3008
+	select SND_SOC_PCM3168A_I2C if I2C
+	select SND_SOC_PCM3168A_SPI if SPI_MASTER
 	select SND_SOC_PCM512x_I2C if I2C
 	select SND_SOC_PCM512x_SPI if SPI_MASTER
 	select SND_SOC_RT286 if I2C
 	select SND_SOC_RT298 if I2C
+	select SND_SOC_RT5616 if I2C
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_RT5640 if I2C
 	select SND_SOC_RT5645 if I2C
 	select SND_SOC_RT5651 if I2C
+	select SND_SOC_RT5659 if I2C
 	select SND_SOC_RT5670 if I2C
 	select SND_SOC_RT5677 if I2C && SPI_MASTER
 	select SND_SOC_SGTL5000 if I2C
@@ -195,10 +203,12 @@ config SND_SOC_88PM860X
 
 config SND_SOC_ARIZONA
 	tristate
+	default y if SND_SOC_CS47L24=y
 	default y if SND_SOC_WM5102=y
 	default y if SND_SOC_WM5110=y
 	default y if SND_SOC_WM8997=y
 	default y if SND_SOC_WM8998=y
+	default m if SND_SOC_CS47L24=m
 	default m if SND_SOC_WM5102=m
 	default m if SND_SOC_WM5110=m
 	default m if SND_SOC_WM8997=m
@@ -211,9 +221,12 @@ config SND_SOC_WM_HUBS
 
 config SND_SOC_WM_ADSP
 	tristate
+	select SND_SOC_COMPRESS
+	default y if SND_SOC_CS47L24=y
 	default y if SND_SOC_WM5102=y
 	default y if SND_SOC_WM5110=y
 	default y if SND_SOC_WM2200=y
+	default m if SND_SOC_CS47L24=m
 	default m if SND_SOC_WM5102=m
 	default m if SND_SOC_WM5110=m
 	default m if SND_SOC_WM2200=m
@@ -422,6 +435,9 @@ config SND_SOC_CS4349
 	tristate "Cirrus Logic CS4349 CODEC"
 	depends on I2C
 
+config SND_SOC_CS47L24
+	tristate
+
 config SND_SOC_CX20442
 	tristate
 	depends on TTY
@@ -439,6 +455,9 @@ config SND_SOC_DA7210
 config SND_SOC_DA7213
         tristate
 
+config SND_SOC_DA7218
+	tristate
+
 config SND_SOC_DA7219
         tristate
 
@@ -468,9 +487,17 @@ config SND_SOC_ES8328_SPI
 config SND_SOC_GTM601
 	tristate 'GTM601 UMTS modem audio codec'
 
+config SND_SOC_HDAC_HDMI
+	tristate
+	select SND_HDA_EXT_CORE
+	select HDMI
+
 config SND_SOC_ICS43432
 	tristate
 
+config SND_SOC_INNO_RK3036
+	tristate "Inno codec driver for RK3036 SoC"
+
 config SND_SOC_ISABELLE
         tristate
 
@@ -506,6 +533,21 @@ config SND_SOC_PCM1792A
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_PCM3168A
+	tristate
+
+config SND_SOC_PCM3168A_I2C
+	tristate "Texas Instruments PCM3168A CODEC - I2C"
+	depends on I2C
+	select SND_SOC_PCM3168A
+	select REGMAP_I2C
+
+config SND_SOC_PCM3168A_SPI
+	tristate "Texas Instruments PCM3168A CODEC - SPI"
+	depends on SPI_MASTER
+	select SND_SOC_PCM3168A
+	select REGMAP_SPI
+
 config SND_SOC_PCM512x
 	tristate
 
@@ -523,14 +565,18 @@ config SND_SOC_PCM512x_SPI
 
 config SND_SOC_RL6231
 	tristate
+	default y if SND_SOC_RT5616=y
 	default y if SND_SOC_RT5640=y
 	default y if SND_SOC_RT5645=y
 	default y if SND_SOC_RT5651=y
+	default y if SND_SOC_RT5659=y
 	default y if SND_SOC_RT5670=y
 	default y if SND_SOC_RT5677=y
+	default m if SND_SOC_RT5616=m
 	default m if SND_SOC_RT5640=m
 	default m if SND_SOC_RT5645=m
 	default m if SND_SOC_RT5651=m
+	default m if SND_SOC_RT5659=m
 	default m if SND_SOC_RT5670=m
 	default m if SND_SOC_RT5677=m
 
@@ -549,6 +595,9 @@ config SND_SOC_RT298
 	tristate
 	depends on I2C
 
+config SND_SOC_RT5616
+	tristate
+
 config SND_SOC_RT5631
 	tristate "Realtek ALC5631/RT5631 CODEC"
 	depends on I2C
@@ -562,6 +611,9 @@ config SND_SOC_RT5645
 config SND_SOC_RT5651
 	tristate
 
+config SND_SOC_RT5659
+	tristate
+
 config SND_SOC_RT5670
 	tristate
 
@@ -838,7 +890,8 @@ config SND_SOC_WM8971
 	tristate
 
 config SND_SOC_WM8974
-	tristate
+	tristate "Wolfson Microelectronics WM8974 codec"
+	depends on I2C
 
 config SND_SOC_WM8978
 	tristate "Wolfson Microelectronics WM8978 codec"
@@ -891,6 +944,7 @@ config SND_SOC_WM9712
 
 config SND_SOC_WM9713
 	tristate
+	select REGMAP_AC97
 
 # Amp
 config SND_SOC_LM4857

sound/soc/codecs/Makefile

@@ -47,9 +47,11 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o
 snd-soc-cs42xx8-objs := cs42xx8.o
 snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
 snd-soc-cs4349-objs := cs4349.o
+snd-soc-cs47l24-objs := cs47l24.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
 snd-soc-da7213-objs := da7213.o
+snd-soc-da7218-objs := da7218.o
 snd-soc-da7219-objs := da7219.o da7219-aad.o
 snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
@@ -59,7 +61,9 @@ snd-soc-es8328-objs := es8328.o
 snd-soc-es8328-i2c-objs := es8328-i2c.o
 snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-gtm601-objs := gtm601.o
+snd-soc-hdac-hdmi-objs := hdac_hdmi.o
 snd-soc-ics43432-objs := ics43432.o
+snd-soc-inno-rk3036-objs := inno_rk3036.o
 snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
@@ -78,6 +82,9 @@ snd-soc-nau8825-objs := nau8825.o
 snd-soc-pcm1681-objs := pcm1681.o
 snd-soc-pcm1792a-codec-objs := pcm1792a.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm3168a-objs := pcm3168a.o
+snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
+snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
 snd-soc-pcm512x-objs := pcm512x.o
 snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
 snd-soc-pcm512x-spi-objs := pcm512x-spi.o
@@ -85,10 +92,12 @@ snd-soc-rl6231-objs := rl6231.o
 snd-soc-rl6347a-objs := rl6347a.o
 snd-soc-rt286-objs := rt286.o
 snd-soc-rt298-objs := rt298.o
+snd-soc-rt5616-objs := rt5616.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-rt5640-objs := rt5640.o
 snd-soc-rt5645-objs := rt5645.o
 snd-soc-rt5651-objs := rt5651.o
+snd-soc-rt5659-objs := rt5659.o
 snd-soc-rt5670-objs := rt5670.o
 snd-soc-rt5677-objs := rt5677.o
 snd-soc-rt5677-spi-objs := rt5677-spi.o
@@ -242,9 +251,11 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI)	+= snd-soc-cs4271-spi.o
 obj-$(CONFIG_SND_SOC_CS42XX8)	+= snd-soc-cs42xx8.o
 obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
 obj-$(CONFIG_SND_SOC_CS4349)	+= snd-soc-cs4349.o
+obj-$(CONFIG_SND_SOC_CS47L24)	+= snd-soc-cs47l24.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DA7213)	+= snd-soc-da7213.o
+obj-$(CONFIG_SND_SOC_DA7218)	+= snd-soc-da7218.o
 obj-$(CONFIG_SND_SOC_DA7219)	+= snd-soc-da7219.o
 obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
@@ -254,7 +265,9 @@ obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
 obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
 obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_GTM601)    += snd-soc-gtm601.o
+obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o
 obj-$(CONFIG_SND_SOC_ICS43432)	+= snd-soc-ics43432.o
+obj-$(CONFIG_SND_SOC_INNO_RK3036)	+= snd-soc-inno-rk3036.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
@@ -273,6 +286,9 @@ obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
 obj-$(CONFIG_SND_SOC_PCM1792A)	+= snd-soc-pcm1792a-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM3168A)	+= snd-soc-pcm3168a.o
+obj-$(CONFIG_SND_SOC_PCM3168A_I2C)	+= snd-soc-pcm3168a-i2c.o
+obj-$(CONFIG_SND_SOC_PCM3168A_SPI)	+= snd-soc-pcm3168a-spi.o
 obj-$(CONFIG_SND_SOC_PCM512x)	+= snd-soc-pcm512x.o
 obj-$(CONFIG_SND_SOC_PCM512x_I2C)	+= snd-soc-pcm512x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
@@ -280,10 +296,12 @@ obj-$(CONFIG_SND_SOC_RL6231)	+= snd-soc-rl6231.o
 obj-$(CONFIG_SND_SOC_RL6347A)	+= snd-soc-rl6347a.o
 obj-$(CONFIG_SND_SOC_RT286)	+= snd-soc-rt286.o
 obj-$(CONFIG_SND_SOC_RT298)	+= snd-soc-rt298.o
+obj-$(CONFIG_SND_SOC_RT5616)	+= snd-soc-rt5616.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_RT5640)	+= snd-soc-rt5640.o
 obj-$(CONFIG_SND_SOC_RT5645)	+= snd-soc-rt5645.o
 obj-$(CONFIG_SND_SOC_RT5651)	+= snd-soc-rt5651.o
+obj-$(CONFIG_SND_SOC_RT5659)	+= snd-soc-rt5659.o
 obj-$(CONFIG_SND_SOC_RT5670)	+= snd-soc-rt5670.o
 obj-$(CONFIG_SND_SOC_RT5677)	+= snd-soc-rt5677.o
 obj-$(CONFIG_SND_SOC_RT5677_SPI)	+= snd-soc-rt5677-spi.o

sound/soc/codecs/ak4613.c

@@ -70,18 +70,11 @@
 #define FMT_MASK	(0xf8)
 
 /* CTRL2 */
+#define DFS_MASK		(3 << 2)
 #define DFS_NORMAL_SPEED	(0 << 2)
 #define DFS_DOUBLE_SPEED	(1 << 2)
 #define DFS_QUAD_SPEED		(2 << 2)
 
-struct ak4613_priv {
-	struct mutex lock;
-
-	unsigned int fmt;
-	u8 fmt_ctrl;
-	int cnt;
-};
-
 struct ak4613_formats {
 	unsigned int width;
 	unsigned int fmt;
@@ -92,6 +85,16 @@ struct ak4613_interface {
 	struct ak4613_formats playback;
 };
 
+struct ak4613_priv {
+	struct mutex lock;
+	const struct ak4613_interface *iface;
+
+	unsigned int fmt;
+	u8 oc;
+	u8 ic;
+	int cnt;
+};
+
 /*
  * Playback Volume
  *
@@ -126,7 +129,7 @@ static const struct reg_default ak4613_reg[] = {
 	{ 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
 };
 
-#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE_TO_VAL(fmts) ((fmts - ak4613_iface) << 3)
 #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
 static const struct ak4613_interface ak4613_iface[] = {
 	/* capture */				/* playback */
@@ -240,7 +243,7 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
 		priv->cnt = 0;
 	}
 	if (!priv->cnt)
-		priv->fmt_ctrl = NO_FMT;
+		priv->iface = NULL;
 	mutex_unlock(&priv->lock);
 }
 
@@ -265,13 +268,35 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
+static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface,
+				    int is_play,
+				    unsigned int fmt, unsigned int width)
+{
+	const struct ak4613_formats *fmts;
+
+	fmts = (is_play) ? &iface->playback : &iface->capture;
+
+	if (fmts->fmt != fmt)
+		return false;
+
+	if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+		if (fmts->width != width)
+			return false;
+	} else {
+		if (fmts->width < width)
+			return false;
+	}
+
+	return true;
+}
+
 static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
-	const struct ak4613_formats *fmts;
+	const struct ak4613_interface *iface;
 	struct device *dev = codec->dev;
 	unsigned int width = params_width(params);
 	unsigned int fmt = priv->fmt;
@@ -305,33 +330,27 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
 	 * It doesn't support TDM at this point
 	 */
 	fmt_ctrl = NO_FMT;
-	for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
-		fmts = (is_play) ?	&ak4613_iface[i].playback :
-					&ak4613_iface[i].capture;
-
-		if (fmts->fmt != fmt)
-			continue;
+	ret = -EINVAL;
+	iface = NULL;
 
-		if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
-			if (fmts->width != width)
-				continue;
-		} else {
-			if (fmts->width < width)
+	mutex_lock(&priv->lock);
+	if (priv->iface) {
+		if (ak4613_dai_fmt_matching(priv->iface, is_play, fmt, width))
+			iface = priv->iface;
+	} else {
+		for (i = ARRAY_SIZE(ak4613_iface); i >= 0; i--) {
+			if (!ak4613_dai_fmt_matching(ak4613_iface + i,
+						     is_play,
+						     fmt, width))
 				continue;
+			iface = ak4613_iface + i;
+			break;
 		}
-
-		fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
-		break;
 	}
 
-	ret = -EINVAL;
-	if (fmt_ctrl == NO_FMT)
-		goto hw_params_end;
-
-	mutex_lock(&priv->lock);
-	if ((priv->fmt_ctrl == NO_FMT) ||
-	    (priv->fmt_ctrl == fmt_ctrl)) {
-		priv->fmt_ctrl = fmt_ctrl;
+	if ((priv->iface == NULL) ||
+	    (priv->iface == iface)) {
+		priv->iface = iface;
 		priv->cnt++;
 		ret = 0;
 	}
@@ -340,8 +359,13 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		goto hw_params_end;
 
+	fmt_ctrl = AUDIO_IFACE_TO_VAL(iface);
+
 	snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
-	snd_soc_write(codec, CTRL2, ctrl2);
+	snd_soc_update_bits(codec, CTRL2, DFS_MASK, ctrl2);
+
+	snd_soc_write(codec, ICTRL, priv->ic);
+	snd_soc_write(codec, OCTRL, priv->oc);
 
 hw_params_end:
 	if (ret < 0)
@@ -431,6 +455,28 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
 	.num_dapm_routes	= ARRAY_SIZE(ak4613_intercon),
 };
 
+static void ak4613_parse_of(struct ak4613_priv *priv,
+			    struct device *dev)
+{
+	struct device_node *np = dev->of_node;
+	char prop[32];
+	int i;
+
+	/* Input 1 - 2 */
+	for (i = 0; i < 2; i++) {
+		snprintf(prop, sizeof(prop), "asahi-kasei,in%d-single-end", i + 1);
+		if (!of_get_property(np, prop, NULL))
+			priv->ic |= 1 << i;
+	}
+
+	/* Output 1 - 6 */
+	for (i = 0; i < 6; i++) {
+		snprintf(prop, sizeof(prop), "asahi-kasei,out%d-single-end", i + 1);
+		if (!of_get_property(np, prop, NULL))
+			priv->oc |= 1 << i;
+	}
+}
+
 static int ak4613_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -458,7 +504,9 @@ static int ak4613_i2c_probe(struct i2c_client *i2c,
 	if (!priv)
 		return -ENOMEM;
 
-	priv->fmt_ctrl		= NO_FMT;
+	ak4613_parse_of(priv, dev);
+
+	priv->iface		= NULL;
 	priv->cnt		= 0;
 
 	mutex_init(&priv->lock);

sound/soc/codecs/arizona.c

@@ -310,7 +310,7 @@ int arizona_init_gpio(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(arizona_init_gpio);
 
-const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
+const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"None",
 	"Tone Generator 1",
 	"Tone Generator 2",
@@ -418,7 +418,7 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 };
 EXPORT_SYMBOL_GPL(arizona_mixer_texts);
 
-int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
+unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
 	0x00,  /* None */
 	0x04,  /* Tone */
 	0x05,
@@ -555,12 +555,12 @@ const char *arizona_sample_rate_val_to_name(unsigned int rate_val)
 }
 EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name);
 
-const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
+const char * const arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
 	"SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate",
 };
 EXPORT_SYMBOL_GPL(arizona_rate_text);
 
-const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = {
+const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = {
 	0, 1, 2, 8,
 };
 EXPORT_SYMBOL_GPL(arizona_rate_val);
@@ -702,6 +702,100 @@ const struct soc_enum arizona_in_dmic_osr[] = {
 };
 EXPORT_SYMBOL_GPL(arizona_in_dmic_osr);
 
+static const char * const arizona_anc_input_src_text[] = {
+	"None", "IN1", "IN2", "IN3", "IN4",
+};
+
+static const char * const arizona_anc_channel_src_text[] = {
+	"None", "Left", "Right", "Combine",
+};
+
+const struct soc_enum arizona_anc_input_src[] = {
+	SOC_ENUM_SINGLE(ARIZONA_ANC_SRC,
+			ARIZONA_IN_RXANCL_SEL_SHIFT,
+			ARRAY_SIZE(arizona_anc_input_src_text),
+			arizona_anc_input_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_FCL_ADC_REFORMATTER_CONTROL,
+			ARIZONA_FCL_MIC_MODE_SEL,
+			ARRAY_SIZE(arizona_anc_channel_src_text),
+			arizona_anc_channel_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_ANC_SRC,
+			ARIZONA_IN_RXANCR_SEL_SHIFT,
+			ARRAY_SIZE(arizona_anc_input_src_text),
+			arizona_anc_input_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_FCR_ADC_REFORMATTER_CONTROL,
+			ARIZONA_FCR_MIC_MODE_SEL,
+			ARRAY_SIZE(arizona_anc_channel_src_text),
+			arizona_anc_channel_src_text),
+};
+EXPORT_SYMBOL_GPL(arizona_anc_input_src);
+
+static const char * const arizona_anc_ng_texts[] = {
+	"None",
+	"Internal",
+	"External",
+};
+
+SOC_ENUM_SINGLE_DECL(arizona_anc_ng_enum, SND_SOC_NOPM, 0,
+		     arizona_anc_ng_texts);
+EXPORT_SYMBOL_GPL(arizona_anc_ng_enum);
+
+static const char * const arizona_output_anc_src_text[] = {
+	"None", "RXANCL", "RXANCR",
+};
+
+const struct soc_enum arizona_output_anc_src[] = {
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
+			ARIZONA_OUT1L_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1R,
+			ARIZONA_OUT1R_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L,
+			ARIZONA_OUT2L_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2R,
+			ARIZONA_OUT2R_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
+			ARIZONA_OUT3L_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_DAC_VOLUME_LIMIT_3R,
+			ARIZONA_OUT3R_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_4L,
+			ARIZONA_OUT4L_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_4R,
+			ARIZONA_OUT4R_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_5L,
+			ARIZONA_OUT5L_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_5R,
+			ARIZONA_OUT5R_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_6L,
+			ARIZONA_OUT6L_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+	SOC_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_6R,
+			ARIZONA_OUT6R_ANC_SRC_SHIFT,
+			ARRAY_SIZE(arizona_output_anc_src_text),
+			arizona_output_anc_src_text),
+};
+EXPORT_SYMBOL_GPL(arizona_output_anc_src);
+
 static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
 {
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
@@ -1023,6 +1117,31 @@ void arizona_init_dvfs(struct arizona_priv *priv)
 }
 EXPORT_SYMBOL_GPL(arizona_init_dvfs);
 
+int arizona_anc_ev(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol,
+		   int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	unsigned int mask = 0x3 << w->shift;
+	unsigned int val;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		val = 1 << w->shift;
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		val = 1 << (w->shift + 1);
+		break;
+	default:
+		return 0;
+	}
+
+	snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_anc_ev);
+
 static unsigned int arizona_opclk_ref_48k_rates[] = {
 	6144000,
 	12288000,
@@ -1095,7 +1214,7 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 	unsigned int reg;
 	unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK;
 	unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT;
-	unsigned int *clk;
+	int *clk;
 
 	switch (clk_id) {
 	case ARIZONA_CLK_SYSCLK:
@@ -1901,18 +2020,18 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 	}
 
 	switch (fll->arizona->type) {
+	case WM5102:
+	case WM8997:
+		return init_ratio;
 	case WM5110:
 	case WM8280:
 		if (fll->arizona->rev < 3 || sync)
 			return init_ratio;
 		break;
-	case WM8998:
-	case WM1814:
+	default:
 		if (sync)
 			return init_ratio;
 		break;
-	default:
-		return init_ratio;
 	}
 
 	cfg->fratio = init_ratio - 1;
@@ -2093,9 +2212,9 @@ static int arizona_enable_fll(struct arizona_fll *fll)
 		/* Facilitate smooth refclk across the transition */
 		regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9,
 					 ARIZONA_FLL1_GAIN_MASK, 0);
-		regmap_update_bits_async(fll->arizona->regmap, fll->base + 1,
-					 ARIZONA_FLL1_FREERUN,
-					 ARIZONA_FLL1_FREERUN);
+		regmap_update_bits(fll->arizona->regmap, fll->base + 1,
+				   ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN);
+		udelay(32);
 	}
 
 	/*

sound/soc/codecs/arizona.h

@@ -57,7 +57,7 @@
 #define ARIZONA_CLK_98MHZ  5
 #define ARIZONA_CLK_147MHZ 6
 
-#define ARIZONA_MAX_DAI  6
+#define ARIZONA_MAX_DAI  8
 #define ARIZONA_MAX_ADSP 4
 
 #define ARIZONA_DVFS_SR1_RQ	0x001
@@ -96,8 +96,8 @@ struct arizona_priv {
 #define ARIZONA_NUM_MIXER_INPUTS 104
 
 extern const unsigned int arizona_mixer_tlv[];
-extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
-extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
+extern const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
+extern unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 
 #define ARIZONA_GAINMUX_CONTROLS(name, base) \
 	SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1,		\
@@ -216,8 +216,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 #define ARIZONA_RATE_ENUM_SIZE 4
 #define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14
 
-extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
-extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
+extern const char * const arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
+extern const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
 extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
 extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
 
@@ -242,6 +242,10 @@ extern const struct soc_enum arizona_in_dmic_osr[];
 
 extern const struct snd_kcontrol_new arizona_adsp2_rate_controls[];
 
+extern const struct soc_enum arizona_anc_input_src[];
+extern const struct soc_enum arizona_anc_ng_enum;
+extern const struct soc_enum arizona_output_anc_src[];
+
 extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol,
 			 int event);
@@ -251,6 +255,9 @@ extern int arizona_out_ev(struct snd_soc_dapm_widget *w,
 extern int arizona_hp_ev(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol,
 			 int event);
+extern int arizona_anc_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event);
 
 extern int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol);

sound/soc/codecs/cs47l24.h

+/*
+ * cs47l24.h  --  ALSA SoC Audio driver for Cirrus Logic CS47L24
+ *
+ * Copyright 2015 Cirrus Logic Inc.
+ *
+ * Author: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _CS47L24_H
+#define _CS47L24_H
+
+#include "arizona.h"
+
+#define CS47L24_FLL1        1
+#define CS47L24_FLL2        2
+#define CS47L24_FLL1_REFCLK 3
+#define CS47L24_FLL2_REFCLK 4
+
+#endif

sound/soc/codecs/da7219.c

@@ -968,10 +968,11 @@ static const struct snd_soc_dapm_route da7219_audio_map[] = {
 	{"Mixin PGA", NULL, "Mic PGA"},
 	{"ADC", NULL, "Mixin PGA"},
 
-	{"Sidetone Filter", NULL, "ADC"},
 	{"Mixer In", NULL, "Mixer In Supply"},
 	{"Mixer In", "Mic Switch", "ADC"},
 
+	{"Sidetone Filter", NULL, "Mixer In"},
+
 	{"Tone Generator", NULL, "TONE"},
 
 	DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"),
@@ -1073,11 +1074,8 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	u32 freq_ref;
 	u64 frac_div;
 
-	/* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
-	if (da7219->mclk_rate == 32768) {
-		indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
-		indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
-	} else if (da7219->mclk_rate < 2000000) {
+	/* Verify 2MHz - 54MHz MCLK provided, and set input divider */
+	if (da7219->mclk_rate < 2000000) {
 		dev_err(codec->dev, "PLL input clock %d below valid range\n",
 			da7219->mclk_rate);
 		return -EINVAL;
@@ -1118,9 +1116,6 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	case DA7219_SYSCLK_PLL_SRM:
 		pll_ctrl |= DA7219_PLL_MODE_SRM;
 		break;
-	case DA7219_SYSCLK_PLL_32KHZ:
-		pll_ctrl |= DA7219_PLL_MODE_32KHZ;
-		break;
 	default:
 		dev_err(codec->dev, "Invalid PLL config\n");
 		return -EINVAL;
@@ -1306,7 +1301,7 @@ static int da7219_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	channels = params_channels(params);
-	if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) {
+	if ((channels < 1) || (channels > DA7219_DAI_CH_NUM_MAX)) {
 		dev_err(codec->dev,
 			"Invalid number of channels, only 1 to %d supported\n",
 			DA7219_DAI_CH_NUM_MAX);
@@ -1405,28 +1400,12 @@ static const struct of_device_id da7219_of_match[] = {
 };
 MODULE_DEVICE_TABLE(of, da7219_of_match);
 
-static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec,
-						 u32 val)
-{
-	switch (val) {
-	case 1050:
-		return DA7219_LDO_LVL_SEL_1_05V;
-	case 1100:
-		return DA7219_LDO_LVL_SEL_1_10V;
-	case 1200:
-		return DA7219_LDO_LVL_SEL_1_20V;
-	case 1400:
-		return DA7219_LDO_LVL_SEL_1_40V;
-	default:
-		dev_warn(codec->dev, "Invalid LDO level");
-		return DA7219_LDO_LVL_SEL_1_05V;
-	}
-}
-
 static enum da7219_micbias_voltage
 	da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
 {
 	switch (val) {
+	case 1600:
+		return DA7219_MICBIAS_1_6V;
 	case 1800:
 		return DA7219_MICBIAS_1_8V;
 	case 2000:
@@ -1469,9 +1448,6 @@ static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec)
 	if (!pdata)
 		return NULL;
 
-	if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0)
-		pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32);
-
 	if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0)
 		pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32);
 	else
@@ -1516,24 +1492,13 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec,
 			snd_soc_update_bits(codec, DA7219_REFERENCES,
 					    DA7219_BIAS_EN_MASK,
 					    DA7219_BIAS_EN_MASK);
-
-			/* Enable Internal Digital LDO */
-			snd_soc_update_bits(codec, DA7219_LDO_CTRL,
-					    DA7219_LDO_EN_MASK,
-					    DA7219_LDO_EN_MASK);
 		}
 		break;
 	case SND_SOC_BIAS_OFF:
-		/* Only disable if jack detection not active */
-		if (!da7219->aad->jack) {
-			/* Bypass Internal Digital LDO */
-			snd_soc_update_bits(codec, DA7219_LDO_CTRL,
-					    DA7219_LDO_EN_MASK, 0);
-
-			/* Master bias */
+		/* Only disable master bias if jack detection not active */
+		if (!da7219->aad->jack)
 			snd_soc_update_bits(codec, DA7219_REFERENCES,
 					    DA7219_BIAS_EN_MASK, 0);
-		}
 
 		/* MCLK */
 		if (da7219->mclk)
@@ -1600,21 +1565,9 @@ static void da7219_handle_pdata(struct snd_soc_codec *codec)
 	if (pdata) {
 		u8 micbias_lvl = 0;
 
-		/* Internal LDO */
-		switch (pdata->ldo_lvl_sel) {
-		case DA7219_LDO_LVL_SEL_1_05V:
-		case DA7219_LDO_LVL_SEL_1_10V:
-		case DA7219_LDO_LVL_SEL_1_20V:
-		case DA7219_LDO_LVL_SEL_1_40V:
-			snd_soc_update_bits(codec, DA7219_LDO_CTRL,
-					    DA7219_LDO_LEVEL_SELECT_MASK,
-					    (pdata->ldo_lvl_sel <<
-					     DA7219_LDO_LEVEL_SELECT_SHIFT));
-			break;
-		}
-
 		/* Mic Bias voltages */
 		switch (pdata->micbias_lvl) {
+		case DA7219_MICBIAS_1_6V:
 		case DA7219_MICBIAS_1_8V:
 		case DA7219_MICBIAS_2_0V:
 		case DA7219_MICBIAS_2_2V:
@@ -1662,10 +1615,12 @@ static int da7219_probe(struct snd_soc_codec *codec)
 	/* Check if MCLK provided */
 	da7219->mclk = devm_clk_get(codec->dev, "mclk");
 	if (IS_ERR(da7219->mclk)) {
-		if (PTR_ERR(da7219->mclk) != -ENOENT)
-			return PTR_ERR(da7219->mclk);
-		else
+		if (PTR_ERR(da7219->mclk) != -ENOENT) {
+			ret = PTR_ERR(da7219->mclk);
+			goto err_disable_reg;
+		} else {
 			da7219->mclk = NULL;
+		}
 	}
 
 	/* Default PC counter to free-running */
@@ -1693,7 +1648,16 @@ static int da7219_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK);
 
 	/* Initialise AAD block */
-	return da7219_aad_init(codec);
+	ret = da7219_aad_init(codec);
+	if (ret)
+		goto err_disable_reg;
+
+	return 0;
+
+err_disable_reg:
+	regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies);
+
+	return ret;
 }
 
 static int da7219_remove(struct snd_soc_codec *codec)
@@ -1776,7 +1740,7 @@ static struct reg_default da7219_reg_defaults[] = {
 	{ DA7219_DIG_ROUTING_DAC, 0x32 },
 	{ DA7219_DAI_OFFSET_LOWER, 0x00 },
 	{ DA7219_DAI_OFFSET_UPPER, 0x00 },
-	{ DA7219_REFERENCES, 0x00 },
+	{ DA7219_REFERENCES, 0x08 },
 	{ DA7219_MIXIN_L_SELECT, 0x00 },
 	{ DA7219_MIXIN_L_GAIN, 0x03 },
 	{ DA7219_ADC_L_GAIN, 0x6F },
@@ -1811,7 +1775,6 @@ static struct reg_default da7219_reg_defaults[] = {
 	{ DA7219_CHIP_ID1, 0x23 },
 	{ DA7219_CHIP_ID2, 0x93 },
 	{ DA7219_CHIP_REVISION, 0x00 },
-	{ DA7219_LDO_CTRL, 0x00 },
 	{ DA7219_IO_CTRL, 0x00 },
 	{ DA7219_GAIN_RAMP_CTRL, 0x00 },
 	{ DA7219_PC_COUNT, 0x02 },

sound/soc/codecs/da7219.h

@@ -85,7 +85,6 @@
 #define DA7219_CHIP_ID1			0x81
 #define DA7219_CHIP_ID2			0x82
 #define DA7219_CHIP_REVISION		0x83
-#define DA7219_LDO_CTRL			0x90
 #define DA7219_IO_CTRL			0x91
 #define DA7219_GAIN_RAMP_CTRL		0x92
 #define DA7219_PC_COUNT			0x94
@@ -207,7 +206,6 @@
 #define DA7219_PLL_MODE_BYPASS		(0x0 << 6)
 #define DA7219_PLL_MODE_NORMAL		(0x1 << 6)
 #define DA7219_PLL_MODE_SRM		(0x2 << 6)
-#define DA7219_PLL_MODE_32KHZ		(0x3 << 6)
 
 /* DA7219_PLL_FRAC_TOP = 0x22 */
 #define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT	0
@@ -569,12 +567,6 @@
 #define DA7219_CHIP_MAJOR_SHIFT	4
 #define DA7219_CHIP_MAJOR_MASK	(0xF << 4)
 
-/* DA7219_LDO_CTRL = 0x90 */
-#define DA7219_LDO_LEVEL_SELECT_SHIFT	4
-#define DA7219_LDO_LEVEL_SELECT_MASK	(0x3 << 4)
-#define DA7219_LDO_EN_SHIFT		7
-#define DA7219_LDO_EN_MASK		(0x1 << 7)
-
 /* DA7219_IO_CTRL = 0x91 */
 #define DA7219_IO_VOLTAGE_LEVEL_SHIFT		0
 #define DA7219_IO_VOLTAGE_LEVEL_MASK		(0x1 << 0)
@@ -787,7 +779,6 @@ enum da7219_sys_clk {
 	DA7219_SYSCLK_MCLK = 0,
 	DA7219_SYSCLK_PLL,
 	DA7219_SYSCLK_PLL_SRM,
-	DA7219_SYSCLK_PLL_32KHZ
 };
 
 /* Regulators */

sound/soc/codecs/rt298.c

@@ -854,8 +854,6 @@ static int rt298_set_dai_sysclk(struct snd_soc_dai *dai,
 	} else {
 		snd_soc_update_bits(codec,
 			RT298_I2S_CTRL2, 0x0100, 0x0100);
-		snd_soc_update_bits(codec,
-			RT298_PLL_CTRL, 0x4, 0x4);
 		snd_soc_update_bits(codec,
 			RT298_PLL_CTRL1, 0x20, 0x0);
 	}