Merge tag 'asoc-v4.11' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v4.11

Another release that's mainly focused on drivers rather than core
changes, highlights include:

 - A huge batch of updates to the Intel drivers, mainly around
   DisplayPort and HDMI with some additional board support too.
 - Channel mapping support for HDMI.
 - Support for AllWinner A31 and A33, Everest Semiconductor ES8328,
   Nuvoton NAU8540.

Documentation/devicetree/bindings/misc/atmel-ssc.txt

@@ -24,6 +24,8 @@ Optional properties:
        this parameter to choose where the clock from.
      - By default the clock is from TK pin, if the clock from RK pin, this
        property is needed.
+  - #sound-dai-cells: Should contain <0>.
+     - This property makes the SSC into an automatically registered DAI.
 
 Examples:
 - PDC transfer:

Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt

@@ -2,8 +2,7 @@ Devicetree bindings for the Axentia TSE-850 audio complex
 
 Required properties:
   - compatible: "axentia,tse850-pcm5142"
-  - axentia,ssc-controller: The phandle of the atmel SSC controller used as
-    cpu dai.
+  - axentia,cpu-dai: The phandle of the cpu dai.
   - axentia,audio-codec: The phandle of the PCM5142 codec.
   - axentia,add-gpios: gpio specifier that controls the mixer.
   - axentia,loop1-gpios: gpio specifier that controls loop relays on channel 1.
@@ -43,6 +42,12 @@ the PCM5142 codec.
 
 Example:
 
+	&ssc0 {
+		#sound-dai-cells = <0>;
+
+		status = "okay";
+	};
+
 	&i2c {
 		codec: pcm5142@4c {
 			compatible = "ti,pcm5142";
@@ -77,7 +82,7 @@ Example:
 	sound {
 		compatible = "axentia,tse850-pcm5142";
 
-		axentia,ssc-controller = <&ssc0>;
+		axentia,cpu-dai = <&ssc0>;
 		axentia,audio-codec = <&codec>;
 
 		axentia,add-gpios = <&pioA 8 GPIO_ACTIVE_LOW>;

Documentation/devicetree/bindings/sound/es8328.txt

@@ -4,7 +4,7 @@ This device supports both I2C and SPI.
 
 Required properties:
 
-  - compatible : "everest,es8328"
+  - compatible  : Should be "everest,es8328" or "everest,es8388"
   - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V
   - AVDD-supply : Regulator providing analog supply voltage 3.3V
   - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V

Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt

@@ -4,6 +4,7 @@ Required properties:
 - compatible = "mediatek,mt2701-audio";
 - reg: register location and size
 - interrupts: Should contain AFE interrupt
+- power-domains: should define the power domain
 - clock-names: should have these clock names:
 		"infra_sys_audio_clk",
 		"top_audio_mux1_sel",
@@ -58,6 +59,7 @@ Example:
 		      <0 0x112A0000 0 0x20000>;
 		interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>,
 			     <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>;
+		power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>;
 		clocks = <&infracfg CLK_INFRA_AUDIO>,
 			 <&topckgen CLK_TOP_AUD_MUX1_SEL>,
 			 <&topckgen CLK_TOP_AUD_MUX2_SEL>,

Documentation/devicetree/bindings/sound/nau8540.txt

@@ -0,0 +1,16 @@
+NAU85L40 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "nuvoton,nau8540"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+codec: nau8540@1c {
+       compatible = "nuvoton,nau8540";
+       reg = <0x1c>;
+};

Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt

@@ -0,0 +1,36 @@
+ROCKCHIP RK3288 with HDMI and analog audio
+
+Required properties:
+- compatible: "rockchip,rk3288-hdmi-analog"
+- rockchip,model: The user-visible name of this sound complex
+- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's
+  connected to the CODEC
+- rockchip,audio-codec: The phandle of the analog audio codec.
+- rockchip,routing: A list of the connections between audio components.
+		    Each entry is a pair of strings, the first being the
+		    connection's sink, the second being the connection's
+		    source. For this driver the first string should always be
+		    "Analog".
+
+Optionnal properties:
+- rockchip,hp-en-gpios = The phandle of the GPIO that power up/down the
+  headphone (when the analog output is an headphone).
+- rockchip,hp-det-gpios = The phandle of the GPIO that detects the headphone
+  (when the analog output is an headphone).
+- pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+
+sound {
+	compatible = "rockchip,rockchip-audio-es8388";
+	rockchip,model = "Analog audio output";
+	rockchip,i2s-controller = <&i2s>;
+	rockchip,audio-codec = <&es8388>;
+	rockchip,routing = "Analog", "LOUT2",
+			   "Analog", "ROUT2";
+	rockchip,hp-en-gpios = <&gpio8 0 GPIO_ACTIVE_HIGH>;
+	rockchip,hp-det-gpios = <&gpio7 7 GPIO_ACTIVE_HIGH>;
+	pinctrl-names = "default";
+	pinctrl-0 = <&headphone>;
+};
+

Documentation/devicetree/bindings/sound/sun4i-i2s.txt

@@ -7,6 +7,7 @@ Required properties:
 
 - compatible: should be one of the followings
    - "allwinner,sun4i-a10-i2s"
+   - "allwinner,sun6i-a31-i2s"
 - reg: physical base address of the controller and length of memory mapped
   region.
 - interrupts: should contain the I2S interrupt.
@@ -19,6 +20,10 @@ Required properties:
    - "mod" : module clock for the I2S controller
 - #sound-dai-cells : Must be equal to 0
 
+Required properties for the following compatibles:
+	- "allwinner,sun6i-a31-i2s"
+- resets: phandle to the reset line for this codec
+
 Example:
 
 i2s0: i2s@01c22400 {

Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt

@@ -0,0 +1,63 @@
+Allwinner SUN8I audio codec
+------------------------------------
+
+On Sun8i-A33 SoCs, the audio is separated in different parts:
+	  - A DAI driver. It uses the "sun4i-i2s" driver which is
+	  documented here:
+	  Documentation/devicetree/bindings/sound/sun4i-i2s.txt
+	  - An analog part of the codec which is handled as PRCM registers.
+	  See Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt
+	  - An digital part of the codec which is documented in this current
+	  binding documentation.
+	  - And finally, an audio card which links all the above components.
+	  The simple-audio card will be used.
+	  See Documentation/devicetree/bindings/sound/simple-card.txt
+
+This bindings documentation exposes Sun8i codec (digital part).
+
+Required properties:
+- compatible: must be "allwinner,sun8i-a33-codec"
+- reg: must contain the registers location and length
+- interrupts: must contain the codec interrupt
+- clocks: a list of phandle + clock-specifer pairs, one for each entry
+  in clock-names.
+- clock-names: should contain followings:
+   - "bus": the parent APB clock for this controller
+   - "mod": the parent module clock
+
+Here is an example to add a sound card and the codec binding on sun8i SoCs that
+are similar to A33 using simple-card:
+
+	sound {
+		compatible = "simple-audio-card";
+		simple-audio-card,name = "sun8i-a33-audio";
+		simple-audio-card,format = "i2s";
+		simple-audio-card,frame-master = <&link_codec>;
+		simple-audio-card,bitclock-master = <&link_codec>;
+		simple-audio-card,mclk-fs = <512>;
+		simple-audio-card,aux-devs = <&codec_analog>;
+		simple-audio-card,routing =
+				"Left DAC", "Digital Left DAC",
+				"Right DAC", "Digital Right DAC";
+
+		simple-audio-card,cpu {
+			sound-dai = <&dai>;
+		};
+
+		link_codec: simple-audio-card,codec {
+			sound-dai = <&codec>;
+		};
+
+	soc@01c00000 {
+		[...]
+
+		audio-codec@1c22e00 {
+			#sound-dai-cells = <0>;
+			compatible = "allwinner,sun8i-a33-codec";
+			reg = <0x01c22e00 0x400>;
+			interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>;
+			clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>;
+			clock-names = "bus", "mod";
+		};
+	};
+

Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt

@@ -10,6 +10,7 @@ Required properties:
   - compatible		: should be one of the following:
     - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC
     - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC
+    - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC
 
   - reg			: Offset and length of the register set for the device.
 

Documentation/devicetree/bindings/sound/zte,zx-i2s.txt

@@ -1,10 +1,12 @@
 ZTE ZX296702 I2S controller
 
 Required properties:
- - compatible : Must be "zte,zx296702-i2s"
+ - compatible : Must be one of:
+	"zte,zx296718-i2s", "zte,zx296702-i2s"
+	"zte,zx296702-i2s"
  - reg : Must contain I2S core's registers location and length
  - clocks : Pairs of phandle and specifier referencing the controller's clocks.
- - clock-names: "tx" for the clock to the I2S interface.
+ - clock-names: "wclk" for the wclk, "pclk" for the pclk to the I2S interface.
  - dmas: Pairs of phandle and specifier for the DMA channel that is used by
    the core. The core expects two dma channels for transmit.
  - dma-names : Must be "tx" and "rx"
@@ -16,12 +18,12 @@ please check:
 	* dma/dma.txt
 
 Example:
-	i2s0: i2s0@0b005000 {
+	i2s0: i2s@b005000 {
 		#sound-dai-cells = <0>;
-		compatible = "zte,zx296702-i2s";
+		compatible = "zte,zx296718-i2s", "zte,zx296702-i2s";
 		reg = <0x0b005000 0x1000>;
-		clocks = <&lsp0clk ZX296702_I2S0_DIV>;
-		clock-names = "tx";
+		clocks = <&audiocrm AUDIO_I2S0_WCLK>, <&audiocrm AUDIO_I2S0_PCLK>;
+		clock-names = "wclk", "pclk";
 		interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>;
 		dmas = <&dma 5>, <&dma 6>;
 		dma-names = "tx", "rx";

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

@@ -106,9 +106,7 @@ static struct s3c_audio_pdata i2sv4_pdata = {
 	.dma_playback = DMACH_HSI_I2SV40_TX,
 	.dma_capture = DMACH_HSI_I2SV40_RX,
 	.type = {
-		.i2s = {
-			.quirks = QUIRK_PRI_6CHAN,
-		},
+		.quirks = QUIRK_PRI_6CHAN,
 	},
 };
 

drivers/misc/atmel-ssc.c

@@ -20,6 +20,8 @@
 
 #include <linux/of.h>
 
+#include "../../sound/soc/atmel/atmel_ssc_dai.h"
+
 /* Serialize access to ssc_list and user count */
 static DEFINE_SPINLOCK(user_lock);
 static LIST_HEAD(ssc_list);
@@ -145,6 +147,49 @@ static inline const struct atmel_ssc_platform_data * __init
 		platform_get_device_id(pdev)->driver_data;
 }
 
+#ifdef CONFIG_SND_ATMEL_SOC_SSC
+static int ssc_sound_dai_probe(struct ssc_device *ssc)
+{
+	struct device_node *np = ssc->pdev->dev.of_node;
+	int ret;
+	int id;
+
+	ssc->sound_dai = false;
+
+	if (!of_property_read_bool(np, "#sound-dai-cells"))
+		return 0;
+
+	id = of_alias_get_id(np, "ssc");
+	if (id < 0)
+		return id;
+
+	ret = atmel_ssc_set_audio(id);
+	ssc->sound_dai = !ret;
+
+	return ret;
+}
+
+static void ssc_sound_dai_remove(struct ssc_device *ssc)
+{
+	if (!ssc->sound_dai)
+		return;
+
+	atmel_ssc_put_audio(of_alias_get_id(ssc->pdev->dev.of_node, "ssc"));
+}
+#else
+static inline int ssc_sound_dai_probe(struct ssc_device *ssc)
+{
+	if (of_property_read_bool(ssc->pdev->dev.of_node, "#sound-dai-cells"))
+		return -ENOTSUPP;
+
+	return 0;
+}
+
+static inline void ssc_sound_dai_remove(struct ssc_device *ssc)
+{
+}
+#endif
+
 static int ssc_probe(struct platform_device *pdev)
 {
 	struct resource *regs;
@@ -204,6 +249,9 @@ static int ssc_probe(struct platform_device *pdev)
 	dev_info(&pdev->dev, "Atmel SSC device at 0x%p (irq %d)\n",
 			ssc->regs, ssc->irq);
 
+	if (ssc_sound_dai_probe(ssc))
+		dev_err(&pdev->dev, "failed to auto-setup ssc for audio\n");
+
 	return 0;
 }
 
@@ -211,6 +259,8 @@ static int ssc_remove(struct platform_device *pdev)
 {
 	struct ssc_device *ssc = platform_get_drvdata(pdev);
 
+	ssc_sound_dai_remove(ssc);
+
 	spin_lock(&user_lock);
 	list_del(&ssc->list);
 	spin_unlock(&user_lock);

include/drm/drm_edid.h

@@ -248,6 +248,7 @@ struct detailed_timing {
 # define DRM_ELD_AUD_SYNCH_DELAY_MAX	0xfa	/* 500 ms */
 
 #define DRM_ELD_SPEAKER			7
+# define DRM_ELD_SPEAKER_MASK		0x7f
 # define DRM_ELD_SPEAKER_RLRC		(1 << 6)
 # define DRM_ELD_SPEAKER_FLRC		(1 << 5)
 # define DRM_ELD_SPEAKER_RC		(1 << 4)
@@ -413,6 +414,18 @@ static inline int drm_eld_size(const uint8_t *eld)
 	return DRM_ELD_HEADER_BLOCK_SIZE + eld[DRM_ELD_BASELINE_ELD_LEN] * 4;
 }
 
+/**
+ * drm_eld_get_spk_alloc - Get speaker allocation
+ * @eld: pointer to an ELD memory structure
+ *
+ * The returned value is the speakers mask. User has to use %DRM_ELD_SPEAKER
+ * field definitions to identify speakers.
+ */
+static inline u8 drm_eld_get_spk_alloc(const uint8_t *eld)
+{
+	return eld[DRM_ELD_SPEAKER] & DRM_ELD_SPEAKER_MASK;
+}
+
 /**
  * drm_eld_get_conn_type - Get device type hdmi/dp connected
  * @eld: pointer to an ELD memory structure

include/linux/atmel-ssc.h

@@ -20,6 +20,7 @@ struct ssc_device {
 	int			user;
 	int			irq;
 	bool			clk_from_rk_pin;
+	bool			sound_dai;
 };
 
 struct ssc_device * __must_check ssc_request(unsigned int ssc_num);

include/linux/platform_data/asoc-s3c.h

@@ -18,7 +18,7 @@
 
 extern void s3c64xx_ac97_setup_gpio(int);
 
-struct samsung_i2s {
+struct samsung_i2s_type {
 /* If the Primary DAI has 5.1 Channels */
 #define QUIRK_PRI_6CHAN		(1 << 0)
 /* If the I2S block has a Stereo Overlay Channel */
@@ -47,7 +47,5 @@ struct s3c_audio_pdata {
 	void *dma_capture;
 	void *dma_play_sec;
 	void *dma_capture_mic;
-	union {
-		struct samsung_i2s i2s;
-	} type;
+	struct samsung_i2s_type type;
 };

include/sound/dmaengine_pcm.h

@@ -71,6 +71,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * @slave_id: Slave requester id for the DMA channel.
  * @filter_data: Custom DMA channel filter data, this will usually be used when
  * requesting the DMA channel.
+ * @chan_name: Custom channel name to use when requesting DMA channel.
  * @fifo_size: FIFO size of the DAI controller in bytes
  * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now
  */
@@ -80,6 +81,7 @@ struct snd_dmaengine_dai_dma_data {
 	u32 maxburst;
 	unsigned int slave_id;
 	void *filter_data;
+	const char *chan_name;
 	unsigned int fifo_size;
 	unsigned int flags;
 };
@@ -105,6 +107,10 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
  * playback.
  */
 #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
+/*
+ * The PCM streams have custom channel names specified.
+ */
+#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4)
 
 /**
  * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM

include/sound/simple_card_utils.h

@@ -34,11 +34,12 @@ int asoc_simple_card_set_dailink_name(struct device *dev,
 int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
 				     char *prefix);
 
-#define asoc_simple_card_parse_clk_cpu(node, dai_link, simple_dai)		\
-	asoc_simple_card_parse_clk(node, dai_link->cpu_of_node, simple_dai)
-#define asoc_simple_card_parse_clk_codec(node, dai_link, simple_dai)		\
-	asoc_simple_card_parse_clk(node, dai_link->codec_of_node, simple_dai)
-int asoc_simple_card_parse_clk(struct device_node *node,
+#define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai)		\
+	asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai)
+#define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai)	\
+	asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai)
+int asoc_simple_card_parse_clk(struct device *dev,
+			       struct device_node *node,
 			       struct device_node *dai_of_node,
 			       struct asoc_simple_dai *simple_dai);
 

include/sound/soc-dai.h

@@ -256,6 +256,9 @@ struct snd_soc_dai_driver {
 	int (*resume)(struct snd_soc_dai *dai);
 	/* compress dai */
 	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
+	/* Optional Callback used at pcm creation*/
+	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
+		       struct snd_soc_dai *dai);
 	/* DAI is also used for the control bus */
 	bool bus_control;
 

include/sound/soc.h

@@ -497,6 +497,8 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt);
 
+int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour);
+
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
 int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
@@ -507,9 +509,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
 	const struct snd_pcm_hardware *hw);
 
-int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
-		int cmd, struct snd_soc_platform *platform);
-
 int soc_dai_hw_params(struct snd_pcm_substream *substream,
 		      struct snd_pcm_hw_params *params,
 		      struct snd_soc_dai *dai);
@@ -785,6 +784,10 @@ struct snd_soc_component_driver {
 	int (*suspend)(struct snd_soc_component *);
 	int (*resume)(struct snd_soc_component *);
 
+	/* pcm creation and destruction */
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
+	void (*pcm_free)(struct snd_pcm *);
+
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
 				 struct of_phandle_args *args,
@@ -859,6 +862,8 @@ struct snd_soc_component {
 	void (*remove)(struct snd_soc_component *);
 	int (*suspend)(struct snd_soc_component *);
 	int (*resume)(struct snd_soc_component *);
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
+	void (*pcm_free)(struct snd_pcm *);
 
 	/* machine specific init */
 	int (*init)(struct snd_soc_component *component);
@@ -941,20 +946,11 @@ struct snd_soc_platform_driver {
 	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
-	/*
-	 * For platform caused delay reporting.
-	 * Optional.
-	 */
-	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
-		struct snd_soc_dai *);
-
 	/* platform stream pcm ops */
 	const struct snd_pcm_ops *ops;
 
 	/* platform stream compress ops */
 	const struct snd_compr_ops *compr_ops;
-
-	int (*bespoke_trigger)(struct snd_pcm_substream *, int);
 };
 
 struct snd_soc_dai_link_component {
@@ -1099,6 +1095,8 @@ struct snd_soc_card {
 	const char *name;
 	const char *long_name;
 	const char *driver_name;
+	char dmi_longname[80];
+
 	struct device *dev;
 	struct snd_card *snd_card;
 	struct module *owner;
@@ -1647,37 +1645,21 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform(
 int snd_soc_util_init(void);
 void snd_soc_util_exit(void);
 
-#define snd_soc_of_parse_card_name(card, propname) \
-	snd_soc_of_parse_card_name_from_node(card, NULL, propname)
-int snd_soc_of_parse_card_name_from_node(struct snd_soc_card *card,
-					 struct device_node *np,
-					 const char *propname);
-#define snd_soc_of_parse_audio_simple_widgets(card, propname)\
-	snd_soc_of_parse_audio_simple_widgets_from_node(card, NULL, propname)
-int snd_soc_of_parse_audio_simple_widgets_from_node(struct snd_soc_card *card,
-						    struct device_node *np,
-						    const char *propname);
-
+int snd_soc_of_parse_card_name(struct snd_soc_card *card,
+			       const char *propname);
+int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
+					  const char *propname);
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
 			      unsigned int *tx_mask,
 			      unsigned int *rx_mask,
 			      unsigned int *slots,
 			      unsigned int *slot_width);
-#define snd_soc_of_parse_audio_prefix(card, codec_conf, of_node, propname) \
-	snd_soc_of_parse_audio_prefix_from_node(card, NULL, codec_conf, \
-						of_node, propname)
-void snd_soc_of_parse_audio_prefix_from_node(struct snd_soc_card *card,
-				   struct device_node *np,
+void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card,
 				   struct snd_soc_codec_conf *codec_conf,
 				   struct device_node *of_node,
 				   const char *propname);
-
-#define snd_soc_of_parse_audio_routing(card, propname) \
-	snd_soc_of_parse_audio_routing_from_node(card, NULL, propname)
-int snd_soc_of_parse_audio_routing_from_node(struct snd_soc_card *card,
-					     struct device_node *np,
-					     const char *propname);
-
+int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
+				   const char *propname);
 unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
 				     const char *prefix,
 				     struct device_node **bitclkmaster,

sound/hda/ext/hdac_ext_stream.c

@@ -128,14 +128,17 @@ void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *ebus,
 {
 	struct hdac_stream *hstream = &stream->hstream;
 	struct hdac_bus *bus = &ebus->bus;
+	u32 val;
+	int mask = AZX_PPCTL_PROCEN(hstream->index);
 
 	spin_lock_irq(&bus->reg_lock);
-	if (decouple)
-		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0,
-				AZX_PPCTL_PROCEN(hstream->index));
-	else
-		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
-					AZX_PPCTL_PROCEN(hstream->index), 0);
+	val = readw(bus->ppcap + AZX_REG_PP_PPCTL) & mask;
+
+	if (decouple && !val)
+		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, mask, mask);
+	else if (!decouple && val)
+		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, mask, 0);
+
 	stream->decoupled = decouple;
 	spin_unlock_irq(&bus->reg_lock);
 }