Merge branch 'for-next' into for-linus

Documentation/DocBook/alsa-driver-api.tmpl

@@ -112,6 +112,8 @@
 !Esound/soc/soc-devres.c
 !Esound/soc/soc-io.c
 !Esound/soc/soc-pcm.c
+!Esound/soc/soc-ops.c
+!Esound/soc/soc-compress.c
      </sect1>
      <sect1><title>ASoC DAPM API</title>
 !Esound/soc/soc-dapm.c

Documentation/DocBook/writing-an-alsa-driver.tmpl

@@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime {
 	struct snd_pcm_hardware hw;
 	struct snd_pcm_hw_constraints hw_constraints;
 
-	/* -- interrupt callbacks -- */
-	void (*transfer_ack_begin)(struct snd_pcm_substream *substream);
-	void (*transfer_ack_end)(struct snd_pcm_substream *substream);
-
 	/* -- timer -- */
 	unsigned int timer_resolution;	/* timer resolution */
 
@@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime {
 	  For the operators (callbacks) of each sound driver, most of
 	these records are supposed to be read-only.  Only the PCM
 	middle-layer changes / updates them.  The exceptions are
-	the hardware description (hw), interrupt callbacks
+	the hardware description (hw) DMA buffer information and the
-	(transfer_ack_xxx), DMA buffer information, and the private
+	private data.  Besides, if you use the standard buffer allocation
-	data.  Besides, if you use the standard buffer allocation
 	method via <function>snd_pcm_lib_malloc_pages()</function>,
 	you don't need to set the DMA buffer information by yourself.
 	</para>
@@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime {
         </para>
 	</section>
 
-	<section id="pcm-interface-runtime-intr">
-	<title>Interrupt Callbacks</title>
-	<para>
-	The field <structfield>transfer_ack_begin</structfield> and
-	<structfield>transfer_ack_end</structfield> are called at
-	the beginning and at the end of
-	<function>snd_pcm_period_elapsed()</function>, respectively. 
-	</para>
-	</section>
-
     </section>
 
     <section id="pcm-interface-operators">

Documentation/devicetree/bindings/sound/ak4613.txt

@@ -0,0 +1,17 @@
+AK4613 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak4613"
+- reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+	ak4613: ak4613@0x10 {
+		compatible = "asahi-kasei,ak4613";
+		reg = <0x10>;
+	};
+};

Documentation/devicetree/bindings/sound/ak4642.txt

@@ -7,7 +7,14 @@ Required properties:
   - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
   - reg : The chip select number on the I2C bus
 
-Example:
+Optional properties:
+
+  - #clock-cells :		common clock binding; shall be set to 0
+  - clocks :			common clock binding; MCKI clock
+  - clock-frequency :		common clock binding; frequency of MCKO
+  - clock-output-names :	common clock binding; MCKO clock name
+
+Example 1:
 
 &i2c {
 	ak4648: ak4648@0x12 {
@@ -15,3 +22,16 @@ Example:
 		reg = <0x12>;
 	};
 };
+
+Example 2:
+
+&i2c {
+	ak4643: codec@12 {
+		compatible = "asahi-kasei,ak4643";
+		reg = <0x12>;
+		#clock-cells = <0>;
+		clocks = <&audio_clock>;
+		clock-frequency = <12288000>;
+		clock-output-names = "ak4643_mcko";
+	};
+};

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

@@ -0,0 +1,52 @@
+* Atmel ClassD driver under ALSA SoC architecture
+
+Required properties:
+- compatible
+	Should be "atmel,sama5d2-classd".
+- reg
+	Should contain ClassD registers location and length.
+- interrupts
+	Should contain the IRQ line for the ClassD.
+- dmas
+	One DMA specifiers as described in atmel-dma.txt and dma.txt files.
+- dma-names
+	Must be "tx".
+- clock-names
+	Tuple listing input clock names.
+	Required elements: "pclk", "gclk" and "aclk".
+- clocks
+	Please refer to clock-bindings.txt.
+
+Optional properties:
+- pinctrl-names, pinctrl-0
+	Please refer to pinctrl-bindings.txt.
+- atmel,model
+	The user-visible name of this sound complex.
+	The default value is "CLASSD".
+- atmel,pwm-type
+	PWM modulation type, "single" or "diff".
+	The default value is "single".
+- atmel,non-overlap-time
+	Set non-overlapping time, the unit is nanosecond(ns).
+	There are four values,
+	<5>, <10>, <15>, <20>, the default value is <10>.
+	Non-overlapping will be disabled if not specified.
+
+Example:
+classd: classd@fc048000 {
+		compatible = "atmel,sama5d2-classd";
+		reg = <0xfc048000 0x100>;
+		interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>;
+		dmas = <&dma0
+			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
+			| AT91_XDMAC_DT_PERID(47))>;
+		dma-names = "tx";
+		clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>;
+		clock-names = "pclk", "gclk", "aclk";
+
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_classd_default>;
+		atmel,model = "classd @ SAMA5D2-Xplained";
+		atmel,pwm-type = "diff";
+		atmel,non-overlap-time = <10>;
+};

Documentation/devicetree/bindings/sound/da7213.txt

@@ -0,0 +1,41 @@
+Dialog Semiconductor DA7213 Audio Codec bindings
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7213"
+- reg: Specifies the I2C slave address
+
+Optional properties:
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1
+	[<1600>, <2200>, <2500>, <3000>]
+- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2
+	[<1600>, <2200>, <2500>, <3000>]
+- dlg,dmic-data-sel : DMIC channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic-samplephase : When to sample audio from DMIC.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic-clkrate : DMIC clock frequency (Hz).
+	[<1500000>, <3000000>]
+
+======
+
+Example:
+
+	codec_i2c: da7213@1a {
+		compatible = "dlg,da7213";
+ 		reg = <0x1a>;
+
+ 		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,micbias1-lvl = <2500>;
+		dlg,micbias2-lvl = <2500>;
+
+		dlg,dmic-data-sel = "lrise_rfall";
+		dlg,dmic-samplephase = "between_clkedge";
+		dlg,dmic-clkrate = <3000000>;
+	};

Documentation/devicetree/bindings/sound/da7219.txt

@@ -0,0 +1,106 @@
+Dialog Semiconductor DA7219 Audio Codec bindings
+
+DA7219 is an audio codec with advanced accessory detect features.
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7219"
+- reg: Specifies the I2C slave address
+
+- interrupt-parent : Specifies the phandle of the interrupt controller to which
+  the IRQs from DA7219 are delivered to.
+- interrupts : IRQ line info for DA7219.
+  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
+   further information relating to interrupt properties)
+
+- 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-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,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>]
+- dlg,mic-amp-in-sel : Mic input source type
+	["diff", "se_p", "se_n"]
+
+======
+
+Child node - 'da7219_aad':
+
+Optional properties:
+- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV).
+	[<2800>, <2900>]
+- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms)
+- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms)
+	[<2>, <5>, <10>, <50>, <100>, <200>, <500>]
+- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms)
+	[<200>, <500>, <750>, <1000>]
+- dlg,jack-ins-deb : Debounce time for jack insertion (ms)
+	[<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>]
+- dlg,jack-det-rate: Jack type detection latency (3/4 pole)
+	["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"]
+- dlg,jack-rem-deb : Debounce time for jack removal (ms)
+	[<1>, <5>, <10>, <20>]
+- dlg,a-d-btn-thr : Impedance threshold between buttons A and D
+	[0x0 - 0xFF]
+- dlg,d-b-btn-thr : Impedance threshold between buttons D and B
+	[0x0 - 0xFF]
+- dlg,b-c-btn-thr : Impedance threshold between buttons B and C
+	[0x0 - 0xFF]
+- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic
+	[0x0 - 0xFF]
+- dlg,btn-avg : Number of 8-bit readings for averaged button measurement
+	[<1>, <2>, <4>, <8>]
+- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement
+	[<1>, <2>, <4>, <8>]
+
+======
+
+Example:
+
+	codec: da7219@1a {
+		compatible = "dlg,da7219";
+		reg = <0x1a>;
+
+		interrupt-parent = <&gpio6>;
+		interrupts = <11 IRQ_TYPE_LEVEL_HIGH>;
+
+		VDD-supply = <&reg_audio>;
+		VDDMIC-supply = <&reg_audio>;
+		VDDIO-supply = <&reg_audio>;
+
+		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,ldo-lvl = <1200>;
+		dlg,micbias-lvl = <2600>;
+		dlg,mic-amp-in-sel = "diff";
+
+		da7219_aad {
+			dlg,btn-cfg = <50>;
+			dlg,mic-det-thr = <500>;
+			dlg,jack-ins-deb = <20>;
+			dlg,jack-det-rate = "32ms_64ms";
+			dlg,jack-rem-deb = <1>;
+
+			dlg,a-d-btn-thr = <0xa>;
+			dlg,d-b-btn-thr = <0x16>;
+			dlg,b-c-btn-thr = <0x21>;
+			dlg,c-mic-btn-thr = <0x3E>;
+
+			dlg,btn-avg = <4>;
+			dlg,adc-1bit-rpt = <1>;
+		};
+	};

Documentation/devicetree/bindings/sound/fsl-asoc-card.txt

@@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit
 from the simplification of a new card support and the capability of the wide
 sample rates support through ASRC.
 
-Note: The card is initially designed for those sound cards who use I2S and
+Note: The card is initially designed for those sound cards who use AC'97, I2S
-      PCM DAI formats. However, it'll be also possible to support those non
+      and PCM DAI formats. However, it'll be also possible to support those non
-      I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
+      AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
-      as the driver has been properly upgraded.
+      long as the driver has been properly upgraded.
 
 
 The compatible list for this generic sound card currently:
+ "fsl,imx-audio-ac97"
+
  "fsl,imx-audio-cs42888"
 
  "fsl,imx-audio-wm8962"

Documentation/devicetree/bindings/sound/nau8825.txt

@@ -0,0 +1,102 @@
+Nuvoton NAU8825 audio codec
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "nuvoton,nau8825"
+
+  - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1).
+
+Optional properties:
+  - nuvoton,jkdet-enable: Enable jack detection via JKDET pin.
+  - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled,
+      otherwise pin in high impedance state.
+  - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down.
+  - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low.
+
+  - nuvoton,vref-impedance: VREF Impedance selection
+      0 - Open
+      1 - 25 kOhm
+      2 - 125 kOhm
+      3 - 2.5 kOhm
+
+  - nuvoton,micbias-voltage: Micbias voltage level.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-threshold-num: Number of buttons supported
+  - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as
+    SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R)
+    where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance.
+    Refer datasheet section 10.2 for more information about threshold calculation.
+
+  - nuvoton,sar-hysteresis: Button impedance measurement hysteresis.
+
+  - nuvoton,sar-voltage: Reference voltage for button impedance measurement.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-compare-time: SAR compare time
+      0 - 500 ns
+      1 - 1 us
+      2 - 2 us
+      3 - 4 us
+
+  - nuvoton,sar-sampling-time: SAR sampling time
+      0 - 2 us
+      1 - 4 us
+      2 - 8 us
+      3 - 16 us
+
+  - nuvoton,short-key-debounce: Button short key press debounce time.
+      0 - 30 ms
+      1 - 50 ms
+      2 - 100 ms
+      3 - 30 ms
+
+  - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
+  - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
+
+  - clocks: list of phandle and clock specifier pairs according to common clock bindings for the
+      clocks described in clock-names
+  - clock-names: should include "mclk" for the MCLK master clock
+
+Example:
+
+  headset: nau8825@1a {
+      compatible = "nuvoton,nau8825";
+      reg = <0x1a>;
+      interrupt-parent = <&gpio>;
+      interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>;
+      nuvoton,jkdet-enable;
+      nuvoton,jkdet-pull-enable;
+      nuvoton,jkdet-pull-up;
+      nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>;
+      nuvoton,vref-impedance = <2>;
+      nuvoton,micbias-voltage = <6>;
+      // Setup 4 buttons impedance according to Android specification
+      nuvoton,sar-threshold-num = <4>;
+      nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+      nuvoton,sar-hysteresis = <1>;
+      nuvoton,sar-voltage = <0>;
+      nuvoton,sar-compare-time = <0>;
+      nuvoton,sar-sampling-time = <0>;
+      nuvoton,short-key-debounce = <2>;
+      nuvoton,jack-insert-debounce = <7>;
+      nuvoton,jack-eject-debounce = <7>;
+
+      clock-names = "mclk";
+      clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
+  };

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

@@ -4,10 +4,12 @@ Required properties:
 - compatible			: "renesas,rcar_sound-<soctype>", fallbacks
 				  "renesas,rcar_sound-gen1" if generation1, and
 				  "renesas,rcar_sound-gen2" if generation2
+				  "renesas,rcar_sound-gen3" if generation3
 				  Examples with soctypes are:
 				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
 				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
 				    - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
+				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1
@@ -30,6 +32,11 @@ Required properties:
 - rcar_sound,dai		: DAI contents.
 				  The number of DAI subnode should be same as HW.
 				  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
+- #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
 
 SSI subnode properties:
 - interrupts			: Should contain SSI interrupt for PIO transfer

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

@@ -12,8 +12,6 @@ Required properties:
 - reg: physical base address of the controller and length of memory mapped
   region.
 - interrupts: should contain the I2S interrupt.
-- #address-cells: should be 1.
-- #size-cells: should be 0.
 - 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".
@@ -21,6 +19,7 @@ Required properties:
 - clock-names: should contain followings:
    - "i2s_hclk": clock for I2S BUS
    - "i2s_clk" : clock for I2S controller
+- rockchip,capture-channels: max capture channels, if not set, 2 channels default.
 
 Example for rk3288 I2S controller:
 
@@ -28,10 +27,9 @@ i2s@ff890000 {
 	compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
 	reg = <0xff890000 0x10000>;
 	interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
-	#address-cells = <1>;
-	#size-cells = <0>;
 	dmas = <&pdma1 0>, <&pdma1 1>;
 	dma-names = "tx", "rx";
 	clock-names = "i2s_hclk", "i2s_clk";
 	clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
+	rockchip,capture-channels = <2>;
 };

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

@@ -0,0 +1,40 @@
+* Rockchip SPDIF transceiver
+
+The S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or
+     "rockchip,rk3066-spdif"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- interrupts: should contain the SPDIF interrupt.
+- dmas: DMA specifiers for tx dma. See the DMA client binding,
+  Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should be "tx"
+- clocks: a list of phandle + clock-specifier pairs, one for each entry
+  in clock-names.
+- clock-names: should contain following:
+   - "hclk": clock for SPDIF controller
+   - "mclk" : clock for SPDIF bus
+
+Required properties on RK3288:
+  - rockchip,grf: the phandle of the syscon node for the general register
+                   file (GRF)
+
+Example for the rk3188 SPDIF controller:
+
+spdif: spdif@0x1011e000 {
+	compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
+	reg = <0x1011e000 0x2000>;
+	interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&dmac1_s 8>;
+	dma-names = "tx";
+	clock-names = "hclk", "mclk";
+	clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>;
+	status = "disabled";
+	#sound-dai-cells = <0>;
+};

Documentation/devicetree/bindings/sound/rt5640.txt

@@ -14,7 +14,8 @@ Optional properties:
 
 - realtek,in1-differential
 - realtek,in2-differential
-  Boolean. Indicate MIC1/2 input are differential, rather than single-ended.
+- realtek,in3-differential
+  Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended.
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
@@ -24,9 +25,11 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640:
   * DMIC2
   * MICBIAS1
   * IN1P
-  * IN1R
+  * IN1N
   * IN2P
-  * IN2R
+  * IN2N
+  * IN3P
+  * IN3N
   * HPOL
   * HPOR
   * LOUTL

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

@@ -0,0 +1,27 @@
+* Allwinner A10 Codec
+
+Required properties:
+- compatible: must be either "allwinner,sun4i-a10-codec" or
+  "allwinner,sun7i-a20-codec"
+- reg: must contain the registers location and length
+- interrupts: must contain the codec interrupt
+- dmas: DMA channels for tx and rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry
+  in clock-names.
+- clock-names: should contain followings:
+   - "apb": the parent APB clock for this controller
+   - "codec": the parent module clock
+
+Example:
+codec: codec@01c22c00 {
+	#sound-dai-cells = <0>;
+	compatible = "allwinner,sun7i-a20-codec";
+	reg = <0x01c22c00 0x40>;
+	interrupts = <0 30 4>;
+	clocks = <&apb0_gates 0>, <&codec_clk>;
+	clock-names = "apb", "codec";
+	dmas = <&dma 0 19>, <&dma 0 19>;
+	dma-names = "rx", "tx";
+};

Documentation/devicetree/bindings/sound/tdm-slot.txt

@@ -4,11 +4,15 @@ This specifies audio DAI's TDM slot.
 
 TDM slot properties:
 dai-tdm-slot-num : Number of slots in use.
-dai-tdm-slot-width :  Width in bits for each slot.
+dai-tdm-slot-width : Width in bits for each slot.
+dai-tdm-slot-tx-mask : Transmit direction slot mask, optional
+dai-tdm-slot-rx-mask : Receive direction slot mask, optional
 
 For instance:
 	dai-tdm-slot-num = <2>;
 	dai-tdm-slot-width = <8>;
+	dai-tdm-slot-tx-mask = <0 1>;
+	dai-tdm-slot-rx-mask = <1 0>;
 
 And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
 to specify a explicit mapping of the channels and the slots. If it's absent
@@ -18,3 +22,8 @@ tx and rx masks.
 For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
 for an active slot as default, and the default active bits are at the LSB of
 the masks.
+
+The explicit masks are given as array of integers, where the first
+number presents bit-0 (LSB), second presents bit-1, etc. Any non zero
+number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask()
+does not do anything, if either mask is set non zero value.

Documentation/sound/alsa/hda_codec.txt

@@ -1,322 +0,0 @@
-Notes on Universal Interface for Intel High Definition Audio Codec
-------------------------------------------------------------------
-
-Takashi Iwai <tiwai@suse.de>
-
-
-[Still a draft version]
-
-
-General
-=======
-
-The snd-hda-codec module supports the generic access function for the
-High Definition (HD) audio codecs.  It's designed to be independent
-from the controller code like ac97 codec module.  The real accessors
-from/to the controller must be implemented in the lowlevel driver.
-
-The structure of this module is similar with ac97_codec module.
-Each codec chip belongs to a bus class which communicates with the
-controller.
-
-
-Initialization of Bus Instance
-==============================
-
-The card driver has to create struct hda_bus at first.  The template
-struct should be filled and passed to the constructor:
-
-struct hda_bus_template {
-	void *private_data;
-	struct pci_dev *pci;
-	const char *modelname;
-	struct hda_bus_ops ops;
-};
-
-The card driver can set and use the private_data field to retrieve its
-own data in callback functions.  The pci field is used when the patch
-needs to check the PCI subsystem IDs, so on.  For non-PCI system, it
-doesn't have to be set, of course.
-The modelname field specifies the board's specific configuration.  The
-string is passed to the codec parser, and it depends on the parser how
-the string is used.
-These fields, private_data, pci and modelname are all optional.
-
-The ops field contains the callback functions as the following:
-
-struct hda_bus_ops {
-	int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
-		       unsigned int verb, unsigned int parm);
-	unsigned int (*get_response)(struct hda_codec *codec);
-	void (*private_free)(struct hda_bus *);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	void (*pm_notify)(struct hda_codec *codec);
-#endif
-};
-
-The command callback is called when the codec module needs to send a
-VERB to the controller.  It's always a single command.
-The get_response callback is called when the codec requires the answer
-for the last command.  These two callbacks are mandatory and have to
-be given.
-The third, private_free callback, is optional.  It's called in the
-destructor to release any necessary data in the lowlevel driver.
-
-The pm_notify callback is available only with
-CONFIG_SND_HDA_POWER_SAVE kconfig.  It's called when the codec needs
-to power up or may power down.  The controller should check the all
-belonging codecs on the bus whether they are actually powered off
-(check codec->power_on), and optionally the driver may power down the
-controller side, too.
-
-The bus instance is created via snd_hda_bus_new().  You need to pass
-the card instance, the template, and the pointer to store the
-resultant bus instance.
-
-int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
-		    struct hda_bus **busp);
-
-It returns zero if successful.  A negative return value means any
-error during creation.
-
-
-Creation of Codec Instance
-==========================
-
-Each codec chip on the board is then created on the BUS instance.
-To create a codec instance, call snd_hda_codec_new().
-
-int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
-		      struct hda_codec **codecp);
-
-The first argument is the BUS instance, the second argument is the
-address of the codec, and the last one is the pointer to store the
-resultant codec instance (can be NULL if not needed).
-
-The codec is stored in a linked list of bus instance.  You can follow
-the codec list like:
-
-	struct hda_codec *codec;
-	list_for_each_entry(codec, &bus->codec_list, list) {
-		...
-	}
-
-The codec isn't initialized at this stage properly.  The
-initialization sequence is called when the controls are built later.
-
-
-Codec Access
-============
-
-To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
-snd_hda_param_read() is for reading parameters.
-For writing a sequence of verbs, use snd_hda_sequence_write().
-
-There are variants of cached read/write, snd_hda_codec_write_cache(),
-snd_hda_sequence_write_cache().  These are used for recording the
-register states for the power-management resume.  When no PM is needed,
-these are equivalent with non-cached version.
-
-To retrieve the number of sub nodes connected to the given node, use
-snd_hda_get_sub_nodes().  The connection list can be obtained via
-snd_hda_get_connections() call.
-
-When an unsolicited event happens, pass the event via
-snd_hda_queue_unsol_event() so that the codec routines will process it
-later.
-
-
-(Mixer) Controls
-================
-
-To create mixer controls of all codecs, call
-snd_hda_build_controls().  It then builds the mixers and does
-initialization stuff on each codec.
-
-
-PCM Stuff
-=========
-
-snd_hda_build_pcms() gives the necessary information to create PCM
-streams.  When it's called, each codec belonging to the bus stores 
-codec->num_pcms and codec->pcm_info fields.  The num_pcms indicates
-the number of elements in pcm_info array.  The card driver is supposed
-to traverse the codec linked list, read the pcm information in
-pcm_info array, and build pcm instances according to them. 
-
-The pcm_info array contains the following record:
-
-/* PCM information for each substream */
-struct hda_pcm_stream {
-	unsigned int substreams;	/* number of substreams, 0 = not exist */
-	unsigned int channels_min;	/* min. number of channels */
-	unsigned int channels_max;	/* max. number of channels */
-	hda_nid_t nid;	/* default NID to query rates/formats/bps, or set up */
-	u32 rates;	/* supported rates */
-	u64 formats;	/* supported formats (SNDRV_PCM_FMTBIT_) */
-	unsigned int maxbps;	/* supported max. bit per sample */
-	struct hda_pcm_ops ops;
-};
-
-/* for PCM creation */
-struct hda_pcm {
-	char *name;
-	struct hda_pcm_stream stream[2];
-};
-
-The name can be passed to snd_pcm_new().  The stream field contains
-the information  for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and
-capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions.  The card driver
-should pass substreams to snd_pcm_new() for the number of substreams
-to create.
-
-The channels_min, channels_max, rates and formats should be copied to
-runtime->hw record.  They and maxbps fields are used also to compute
-the format value for the HDA codec and controller.  Call
-snd_hda_calc_stream_format() to get the format value.
-
-The ops field contains the following callback functions:
-
-struct hda_pcm_ops {
-	int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		    struct snd_pcm_substream *substream);
-	int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		     struct snd_pcm_substream *substream);
-	int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		       unsigned int stream_tag, unsigned int format,
-		       struct snd_pcm_substream *substream);
-	int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		       struct snd_pcm_substream *substream);
-};
-
-All are non-NULL, so you can call them safely without NULL check.
-
-The open callback should be called in PCM open after runtime->hw is
-set up.  It may override some setting and constraints additionally.
-Similarly, the close callback should be called in the PCM close.
-
-The prepare callback should be called in PCM prepare.  This will set
-up the codec chip properly for the operation.  The cleanup should be
-called in hw_free to clean up the configuration.
-
-The caller should check the return value, at least for open and
-prepare callbacks.  When a negative value is returned, some error
-occurred.
-
-
-Proc Files
-==========
-
-Each codec dumps the widget node information in
-/proc/asound/card*/codec#* file.  This information would be really
-helpful for debugging.  Please provide its contents together with the
-bug report.
-
-
-Power Management
-================
-
-It's simple:
-Call snd_hda_suspend() in the PM suspend callback.
-Call snd_hda_resume() in the PM resume callback.
-
-
-Codec Preset (Patch)
-====================
-
-To set up and handle the codec functionality fully, each codec may
-have a codec preset (patch).  It's defined in struct hda_codec_preset:
-
-	struct hda_codec_preset {
-		unsigned int id;
-		unsigned int mask;
-		unsigned int subs;
-		unsigned int subs_mask;
-		unsigned int rev;
-		const char *name;
-		int (*patch)(struct hda_codec *codec);
-	};
-
-When the codec id and codec subsystem id match with the given id and
-subs fields bitwise (with bitmask mask and subs_mask), the callback
-patch is called.  The patch callback should initialize the codec and
-set the codec->patch_ops field.  This is defined as below:
-
-	struct hda_codec_ops {
-		int (*build_controls)(struct hda_codec *codec);
-		int (*build_pcms)(struct hda_codec *codec);
-		int (*init)(struct hda_codec *codec);
-		void (*free)(struct hda_codec *codec);
-		void (*unsol_event)(struct hda_codec *codec, unsigned int res);
-	#ifdef CONFIG_PM
-		int (*suspend)(struct hda_codec *codec, pm_message_t state);
-		int (*resume)(struct hda_codec *codec);
-	#endif
-	#ifdef CONFIG_SND_HDA_POWER_SAVE
-		int (*check_power_status)(struct hda_codec *codec,
-					  hda_nid_t nid);
-	#endif
-	};
-
-The build_controls callback is called from snd_hda_build_controls().
-Similarly, the build_pcms callback is called from
-snd_hda_build_pcms().  The init callback is called after
-build_controls to initialize the hardware.
-The free callback is called as a destructor.
-
-The unsol_event callback is called when an unsolicited event is
-received.
-
-The suspend and resume callbacks are for power management.
-They can be NULL if no special sequence is required.  When the resume
-callback is NULL, the driver calls the init callback and resumes the
-registers from the cache.  If other handling is needed, you'd need to
-write your own resume callback.  There, the amp values can be resumed
-via
-	void snd_hda_codec_resume_amp(struct hda_codec *codec);
-and the other codec registers via
-	void snd_hda_codec_resume_cache(struct hda_codec *codec);
-
-The check_power_status callback is called when the amp value of the
-given widget NID is changed.  The codec code can turn on/off the power
-appropriately from this information.
-
-Each entry can be NULL if not necessary to be called.
-
-
-Generic Parser
-==============
-
-When the device doesn't match with any given presets, the widgets are
-parsed via th generic parser (hda_generic.c).  Its support is
-limited: no multi-channel support, for example.
-
-
-Digital I/O
-===========
-
-Call snd_hda_create_spdif_out_ctls() from the patch to create controls
-related with SPDIF out.
-
-
-Helper Functions
-================
-
-snd_hda_get_codec_name() stores the codec name on the given string.
-
-snd_hda_check_board_config() can be used to obtain the configuration
-information matching with the device.  Define the model string table
-and the table with struct snd_pci_quirk entries (zero-terminated),
-and pass it to the function.  The function checks the modelname given
-as a module parameter, and PCI subsystem IDs.  If the matching entry
-is found, it returns the config field value.
-
-snd_hda_add_new_ctls() can be used to create and add control entries.
-Pass the zero-terminated array of struct snd_kcontrol_new
-
-Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
-used for the entry of struct snd_kcontrol_new.
-
-The input MUX helper callbacks for such a control are provided, too:
-snd_hda_input_mux_info() and snd_hda_input_mux_put().  See
-patch_realtek.c for example.

MAINTAINERS

@@ -3368,6 +3368,7 @@ M:	Support Opensource <support.opensource@diasemi.com>
 W:	http://www.dialog-semiconductor.com/products
 S:	Supported
 F:	Documentation/hwmon/da90??
+F:	Documentation/devicetree/bindings/sound/da[79]*.txt
 F:	drivers/gpio/gpio-da90??.c
 F:	drivers/hwmon/da90??-hwmon.c
 F:	drivers/iio/adc/da91??-*.c

drivers/gpu/drm/i915/i915_dma.c

@@ -832,6 +832,7 @@ int i915_driver_load(struct drm_device *dev, unsigned long flags)
 	mutex_init(&dev_priv->sb_lock);
 	mutex_init(&dev_priv->modeset_restore_lock);
 	mutex_init(&dev_priv->csr_lock);
+	mutex_init(&dev_priv->av_mutex);
 
 	intel_pm_setup(dev);
 

drivers/gpu/drm/i915/i915_drv.h

@@ -1885,6 +1885,11 @@ struct drm_i915_private {
 	/* hda/i915 audio component */
 	struct i915_audio_component *audio_component;
 	bool audio_component_registered;
+	/**
+	 * av_mutex - mutex for audio/video sync
+	 *
+	 */
+	struct mutex av_mutex;
 
 	uint32_t hw_context_size;
 	struct list_head context_list;

drivers/gpu/drm/i915/intel_audio.c

@@ -68,6 +68,31 @@ static const struct {
 	{ 148500, AUD_CONFIG_PIXEL_CLOCK_HDMI_148500 },
 };
 
+/* HDMI N/CTS table */
+#define TMDS_297M 297000
+#define TMDS_296M DIV_ROUND_UP(297000 * 1000, 1001)
+static const struct {
+	int sample_rate;
+	int clock;
+	int n;
+	int cts;
+} aud_ncts[] = {
+	{ 44100, TMDS_296M, 4459, 234375 },
+	{ 44100, TMDS_297M, 4704, 247500 },
+	{ 48000, TMDS_296M, 5824, 281250 },
+	{ 48000, TMDS_297M, 5120, 247500 },
+	{ 32000, TMDS_296M, 5824, 421875 },
+	{ 32000, TMDS_297M, 3072, 222750 },
+	{ 88200, TMDS_296M, 8918, 234375 },
+	{ 88200, TMDS_297M, 9408, 247500 },
+	{ 96000, TMDS_296M, 11648, 281250 },
+	{ 96000, TMDS_297M, 10240, 247500 },
+	{ 176400, TMDS_296M, 17836, 234375 },
+	{ 176400, TMDS_297M, 18816, 247500 },
+	{ 192000, TMDS_296M, 23296, 281250 },
+	{ 192000, TMDS_297M, 20480, 247500 },
+};
+
 /* get AUD_CONFIG_PIXEL_CLOCK_HDMI_* value for mode */
 static u32 audio_config_hdmi_pixel_clock(struct drm_display_mode *mode)
 {
@@ -90,6 +115,45 @@ static u32 audio_config_hdmi_pixel_clock(struct drm_display_mode *mode)
 	return hdmi_audio_clock[i].config;
 }
 
+static int audio_config_get_n(const struct drm_display_mode *mode, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(aud_ncts); i++) {
+		if ((rate == aud_ncts[i].sample_rate) &&
+			(mode->clock == aud_ncts[i].clock)) {
+			return aud_ncts[i].n;
+		}
+	}
+	return 0;
+}
+
+static uint32_t audio_config_setup_n_reg(int n, uint32_t val)
+{
+	int n_low, n_up;
+	uint32_t tmp = val;
+
+	n_low = n & 0xfff;
+	n_up = (n >> 12) & 0xff;
+	tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK);
+	tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) |
+			(n_low << AUD_CONFIG_LOWER_N_SHIFT) |
+			AUD_CONFIG_N_PROG_ENABLE);
+	return tmp;
+}
+
+/* check whether N/CTS/M need be set manually */
+static bool audio_rate_need_prog(struct intel_crtc *crtc,
+				 const struct drm_display_mode *mode)
+{
+	if (((mode->clock == TMDS_297M) ||
+		 (mode->clock == TMDS_296M)) &&
+		intel_pipe_has_type(crtc, INTEL_OUTPUT_HDMI))
+		return true;
+	else
+		return false;
+}
+
 static bool intel_eld_uptodate(struct drm_connector *connector,
 			       int reg_eldv, uint32_t bits_eldv,
 			       int reg_elda, uint32_t bits_elda,
@@ -184,6 +248,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder)
 
 	DRM_DEBUG_KMS("Disable audio codec on pipe %c\n", pipe_name(pipe));
 
+	mutex_lock(&dev_priv->av_mutex);
+
 	/* Disable timestamps */
 	tmp = I915_READ(HSW_AUD_CFG(pipe));
 	tmp &= ~AUD_CONFIG_N_VALUE_INDEX;
@@ -199,6 +265,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder)
 	tmp &= ~AUDIO_ELD_VALID(pipe);
 	tmp &= ~AUDIO_OUTPUT_ENABLE(pipe);
 	I915_WRITE(HSW_AUD_PIN_ELD_CP_VLD, tmp);
+
+	mutex_unlock(&dev_priv->av_mutex);
 }
 
 static void hsw_audio_codec_enable(struct drm_connector *connector,
@@ -208,13 +276,20 @@ static void hsw_audio_codec_enable(struct drm_connector *connector,
 	struct drm_i915_private *dev_priv = connector->dev->dev_private;
 	struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc);
 	enum pipe pipe = intel_crtc->pipe;
+	struct i915_audio_component *acomp = dev_priv->audio_component;
 	const uint8_t *eld = connector->eld;
+	struct intel_digital_port *intel_dig_port =
+		enc_to_dig_port(&encoder->base);
+	enum port port = intel_dig_port->port;
 	uint32_t tmp;
 	int len, i;
+	int n, rate;
 
 	DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n",
 		      pipe_name(pipe), drm_eld_size(eld));
 
+	mutex_lock(&dev_priv->av_mutex);
+
 	/* Enable audio presence detect, invalidate ELD */
 	tmp = I915_READ(HSW_AUD_PIN_ELD_CP_VLD);
 	tmp |= AUDIO_OUTPUT_ENABLE(pipe);
@@ -246,13 +321,32 @@ static void hsw_audio_codec_enable(struct drm_connector *connector,
 	/* Enable timestamps */
 	tmp = I915_READ(HSW_AUD_CFG(pipe));
 	tmp &= ~AUD_CONFIG_N_VALUE_INDEX;
-	tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
 	tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK;
 	if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT))
 		tmp |= AUD_CONFIG_N_VALUE_INDEX;
 	else
 		tmp |= audio_config_hdmi_pixel_clock(mode);
+
+	tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
+	if (audio_rate_need_prog(intel_crtc, mode)) {
+		if (!acomp)
+			rate = 0;
+		else if (port >= PORT_A && port <= PORT_E)
+			rate = acomp->aud_sample_rate[port];
+		else {
+			DRM_ERROR("invalid port: %d\n", port);
+			rate = 0;
+		}
+		n = audio_config_get_n(mode, rate);
+		if (n != 0)
+			tmp = audio_config_setup_n_reg(n, tmp);
+		else
+			DRM_DEBUG_KMS("no suitable N value is found\n");
+	}
+
 	I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+
+	mutex_unlock(&dev_priv->av_mutex);
 }
 
 static void ilk_audio_codec_disable(struct intel_encoder *encoder)
@@ -527,12 +621,91 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev)
 	return ret;
 }
 
+static int i915_audio_component_sync_audio_rate(struct device *dev,
+						int port, int rate)
+{
+	struct drm_i915_private *dev_priv = dev_to_i915(dev);
+	struct drm_device *drm_dev = dev_priv->dev;
+	struct intel_encoder *intel_encoder;
+	struct intel_digital_port *intel_dig_port;
+	struct intel_crtc *crtc;
+	struct drm_display_mode *mode;
+	struct i915_audio_component *acomp = dev_priv->audio_component;
+	enum pipe pipe = -1;
+	u32 tmp;
+	int n;
+
+	/* HSW, BDW SKL need this fix */
+	if (!IS_SKYLAKE(dev_priv) &&
+		!IS_BROADWELL(dev_priv) &&
+		!IS_HASWELL(dev_priv))
+		return 0;
+
+	mutex_lock(&dev_priv->av_mutex);
+	/* 1. get the pipe */
+	for_each_intel_encoder(drm_dev, intel_encoder) {
+		if (intel_encoder->type != INTEL_OUTPUT_HDMI)
+			continue;
+		intel_dig_port = enc_to_dig_port(&intel_encoder->base);
+		if (port == intel_dig_port->port) {
+			crtc = to_intel_crtc(intel_encoder->base.crtc);
+			if (!crtc) {
+				DRM_DEBUG_KMS("%s: crtc is NULL\n", __func__);
+				continue;
+			}
+			pipe = crtc->pipe;
+			break;
+		}
+	}
+
+	if (pipe == INVALID_PIPE) {
+		DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port));
+		mutex_unlock(&dev_priv->av_mutex);
+		return -ENODEV;
+	}
+	DRM_DEBUG_KMS("pipe %c connects port %c\n",
+				  pipe_name(pipe), port_name(port));
+	mode = &crtc->config->base.adjusted_mode;
+
+	/* port must be valid now, otherwise the pipe will be invalid */
+	acomp->aud_sample_rate[port] = rate;
+
+	/* 2. check whether to set the N/CTS/M manually or not */
+	if (!audio_rate_need_prog(crtc, mode)) {
+		tmp = I915_READ(HSW_AUD_CFG(pipe));
+		tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
+		I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+		mutex_unlock(&dev_priv->av_mutex);
+		return 0;
+	}
+
+	n = audio_config_get_n(mode, rate);
+	if (n == 0) {
+		DRM_DEBUG_KMS("Using automatic mode for N value on port %c\n",
+					  port_name(port));
+		tmp = I915_READ(HSW_AUD_CFG(pipe));
+		tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
+		I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+		mutex_unlock(&dev_priv->av_mutex);
+		return 0;
+	}
+
+	/* 3. set the N/CTS/M */
+	tmp = I915_READ(HSW_AUD_CFG(pipe));
+	tmp = audio_config_setup_n_reg(n, tmp);
+	I915_WRITE(HSW_AUD_CFG(pipe), tmp);
+
+	mutex_unlock(&dev_priv->av_mutex);
+	return 0;
+}
+
 static const struct i915_audio_component_ops i915_audio_component_ops = {
 	.owner		= THIS_MODULE,
 	.get_power	= i915_audio_component_get_power,
 	.put_power	= i915_audio_component_put_power,
 	.codec_wake_override = i915_audio_component_codec_wake_override,
 	.get_cdclk_freq	= i915_audio_component_get_cdclk_freq,
+	.sync_audio_rate = i915_audio_component_sync_audio_rate,
 };
 
 static int i915_audio_component_bind(struct device *i915_dev,
@@ -540,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev,
 {
 	struct i915_audio_component *acomp = data;
 	struct drm_i915_private *dev_priv = dev_to_i915(i915_dev);
+	int i;
 
 	if (WARN_ON(acomp->ops || acomp->dev))
 		return -EEXIST;
@@ -547,6 +721,9 @@ static int i915_audio_component_bind(struct device *i915_dev,
 	drm_modeset_lock_all(dev_priv->dev);
 	acomp->ops = &i915_audio_component_ops;
 	acomp->dev = i915_dev;
+	BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS);
+	for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++)
+		acomp->aud_sample_rate[i] = 0;
 	dev_priv->audio_component = acomp;
 	drm_modeset_unlock_all(dev_priv->dev);