Merge branch 'asoc-4.20' into asoc-next

Documentation/devicetree/bindings/sound/adi,adau1977.txt

@@ -0,0 +1,54 @@
+Analog Devices ADAU1977/ADAU1978/ADAU1979
+
+Datasheets:
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
+
+This driver supports both the I2C and SPI bus.
+
+Required properties:
+ - compatible: Should contain one of the following:
+               "adi,adau1977"
+               "adi,adau1978"
+               "adi,adau1979"
+
+ - AVDD-supply: analog power supply for the device, please consult
+                Documentation/devicetree/bindings/regulator/regulator.txt
+
+Optional properties:
+ - reset-gpio:  the reset pin for the chip, for more details consult
+                Documentation/devicetree/bindings/gpio/gpio.txt
+
+ - DVDD-supply: supply voltage for the digital core, please consult
+                Documentation/devicetree/bindings/regulator/regulator.txt
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg:         The i2c address. Value depends on the state of ADDR0
+                and ADDR1, as wired in hardware.
+
+Examples:
+
+	adau1977_spi: adau1977@0 {
+		compatible = "adi,adau1977";
+		spi-max-frequency = <600000>;
+
+		AVDD-supply = <&regulator>;
+		DVDD-supply = <&regulator_digital>;
+
+		reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+	};
+
+	adau1977_i2c: adau1977@11 {
+		compatible = "adi,adau1977";
+		reg = <0x11>;
+
+		AVDD-supply = <&regulator>;
+		DVDD-supply = <&regulator_digital>;
+
+		reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+	};

Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt

@@ -0,0 +1,24 @@
+* Amlogic Audio PDM input
+
+Required properties:
+- compatible: 'amlogic,axg-pdm'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk"   : peripheral clock.
+  * "dclk"   : pdm digital clock
+  * "sysclk" : dsp system clock
+- #sound-dai-cells: must be 0.
+
+Example of PDM on the A113 SoC:
+
+pdm: audio-controller@ff632000 {
+	compatible = "amlogic,axg-pdm";
+	reg = <0x0 0xff632000 0x0 0x34>;
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_PDM>,
+		 <&clkc_audio AUD_CLKID_PDM_DCLK>,
+		 <&clkc_audio AUD_CLKID_PDM_SYSCLK>;
+	clock-names = "pclk", "dclk", "sysclk";
+};

Documentation/devicetree/bindings/sound/cs42l51.txt

@@ -0,0 +1,17 @@
+CS42L51 audio CODEC
+
+Optional properties:
+
+  - clocks : a list of phandles + clock-specifiers, one for each entry in
+    clock-names
+
+  - clock-names : must contain "MCLK"
+
+Example:
+
+cs42l51: cs42l51@4a {
+	compatible = "cirrus,cs42l51";
+	reg = <0x4a>;
+	clocks = <&mclk_prov>;
+	clock-names = "MCLK";
+};

Documentation/devicetree/bindings/sound/maxim,max98088.txt

@@ -0,0 +1,23 @@
+MAX98088 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible: "maxim,max98088" or "maxim,max98089".
+- reg: The I2C address of the device.
+
+Optional properties:
+
+- clocks: the clock provider of MCLK, see ../clock/clock-bindings.txt section
+  "consumer" for more information.
+- clock-names: must be set to "mclk"
+
+Example:
+
+max98089: codec@10 {
+	compatible = "maxim,max98089";
+	reg = <0x10>;
+	clocks = <&clks IMX6QDL_CLK_CKO2>;
+	clock-names = "mclk";
+};

Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt

@@ -0,0 +1,23 @@
+Mikroe-PROTO audio board
+
+Required properties:
+  - compatible: "mikroe,mikroe-proto"
+  - dai-format: Must be "i2s".
+  - i2s-controller: The phandle of the I2S controller.
+  - audio-codec: The phandle of the WM8731 audio codec.
+Optional properties:
+  - model: The user-visible name of this sound complex.
+  - bitclock-master: Indicates dai-link bit clock master; for details see simple-card.txt (1).
+  - frame-master: Indicates dai-link frame master; for details see simple-card.txt (1).
+
+(1) : There must be the same master for both bit and frame clocks.
+
+Example:
+	sound {
+		compatible = "mikroe,mikroe-proto";
+		model = "wm8731 @ sama5d2_xplained";
+		i2s-controller = <&i2s0>;
+		audio-codec = <&wm8731>;
+		dai-format = "i2s";
+        };
+};

Documentation/devicetree/bindings/sound/nau8822.txt

@@ -0,0 +1,16 @@
+NAU8822 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "nuvoton,nau8822"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+codec: nau8822@1a {
+	compatible = "nuvoton,nau8822";
+	reg = <0x1a>;
+};

Documentation/devicetree/bindings/sound/pcm3060.txt

@@ -0,0 +1,17 @@
+PCM3060 audio CODEC
+
+This driver supports both I2C and SPI.
+
+Required properties:
+
+- compatible: "ti,pcm3060"
+
+- reg : the I2C address of the device for I2C, the chip select
+        number for SPI.
+
+Examples:
+
+	pcm3060: pcm3060@46 {
+		 compatible = "ti,pcm3060";
+		 reg = <0x46>;
+	};

Documentation/devicetree/bindings/sound/qcom,q6afe.txt

@@ -49,7 +49,7 @@ configuration of each dai. Must contain the following properties.
 	Usage: required for mi2s interface
 	Value type: <prop-encoded-array>
 	Definition: Must be list of serial data lines used by this dai.
-	should be one or more of the 1-4 sd lines.
+	should be one or more of the 0-3 sd lines.
 
  - qcom,tdm-sync-mode:
 	Usage: required for tdm interface
@@ -137,42 +137,42 @@ q6afe@4 {
 
 		prim-mi2s-rx@16 {
 			reg = <16>;
-			qcom,sd-lines = <1 3>;
+			qcom,sd-lines = <0 2>;
 		};
 
 		prim-mi2s-tx@17 {
 			reg = <17>;
-			qcom,sd-lines = <2>;
+			qcom,sd-lines = <1>;
 		};
 
 		sec-mi2s-rx@18 {
 			reg = <18>;
-			qcom,sd-lines = <1 4>;
+			qcom,sd-lines = <0 3>;
 		};
 
 		sec-mi2s-tx@19 {
 			reg = <19>;
-			qcom,sd-lines = <2>;
+			qcom,sd-lines = <1>;
 		};
 
 		tert-mi2s-rx@20 {
 			reg = <20>;
-			qcom,sd-lines = <2 4>;
+			qcom,sd-lines = <1 3>;
 		};
 
 		tert-mi2s-tx@21 {
 			reg = <21>;
-			qcom,sd-lines = <1>;
+			qcom,sd-lines = <0>;
 		};
 
 		quat-mi2s-rx@22 {
 			reg = <22>;
-			qcom,sd-lines = <1>;
+			qcom,sd-lines = <0>;
 		};
 
 		quat-mi2s-tx@23 {
 			reg = <23>;
-			qcom,sd-lines = <2>;
+			qcom,sd-lines = <1>;
 		};
 	};
 };

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

@@ -340,10 +340,12 @@ Required properties:
 - compatible			: "renesas,rcar_sound-<soctype>", fallbacks
 				  "renesas,rcar_sound-gen1" if generation1, and
 				  "renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
-				  "renesas,rcar_sound-gen3" if generation3
+				  "renesas,rcar_sound-gen3" if generation3 (or RZ/G2)
 				  Examples with soctypes are:
 				    - "renesas,rcar_sound-r8a7743" (RZ/G1M)
+				    - "renesas,rcar_sound-r8a7744" (RZ/G1N)
 				    - "renesas,rcar_sound-r8a7745" (RZ/G1E)
+				    - "renesas,rcar_sound-r8a774a1" (RZ/G2M)
 				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
 				    - "renesas,rcar_sound-r8a7779" (R-Car H1)
 				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
@@ -353,6 +355,7 @@ Required properties:
 				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
 				    - "renesas,rcar_sound-r8a7796" (R-Car M3-W)
 				    - "renesas,rcar_sound-r8a77965" (R-Car M3-N)
+				    - "renesas,rcar_sound-r8a77990" (R-Car E3)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1

Documentation/devicetree/bindings/sound/st,sta32x.txt

@@ -19,6 +19,10 @@ Required properties:
 
 Optional properties:
 
+  - clocks, clock-names: Clock specifier for XTI input clock.
+	If specified, the clock will be enabled when the codec is probed,
+	and disabled when it is removed. The 'clock-names' must be set to 'xti'.
+
   -  st,output-conf: number, Selects the output configuration:
 	0: 2-channel (full-bridge) power, 2-channel data-out
 	1: 2 (half-bridge). 1 (full-bridge) on-board power
@@ -39,6 +43,9 @@ Optional properties:
   -  st,thermal-warning-recover:
 	If present, thermal warning recovery is enabled.
 
+  - st,fault-detect-recovery:
+	If present, fault detect recovery is enabled.
+
   -  st,thermal-warning-adjustment:
 	If present, thermal warning adjustment is enabled.
 
@@ -76,6 +83,8 @@ Example:
 codec: sta32x@38 {
 	compatible = "st,sta32x";
 	reg = <0x1c>;
+	clocks = <&clock>;
+	clock-names = "xti";
 	reset-gpios = <&gpio1 19 0>;
 	power-down-gpios = <&gpio1 16 0>;
 	st,output-conf = /bits/ 8  <0x3>;	// set output to 2-channel

Documentation/devicetree/bindings/sound/st,stm32-sai.txt

@@ -31,7 +31,11 @@ SAI subnodes required properties:
   - reg: Base address and size of SAI sub-block register set.
   - clocks: Must contain one phandle and clock specifier pair
 	for sai_ck which feeds the internal clock generator.
+	If the SAI shares a master clock, with another SAI set as MCLK
+	clock provider, SAI provider phandle must be specified here.
   - clock-names: Must contain "sai_ck".
+	Must also contain "MCLK", if SAI shares a master clock,
+	with a SAI set as MCLK clock provider.
   - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt
   - dma-names: identifier string for each DMA request line
 	"tx": if sai sub-block is configured as playback DAI
@@ -51,6 +55,9 @@ SAI subnodes Optional properties:
 	configured according to protocol defined in related DAI link node,
 	such as i2s, left justified, right justified, dsp and pdm protocols.
 	Note: ac97 protocol is not supported by SAI driver
+   - #clock-cells: should be 0. This property must be present if the SAI device
+	is a master clock provider, according to clocks bindings, described in
+	Documentation/devicetree/bindings/clock/clock-bindings.txt.
 
 The device node should contain one 'port' child node with one child 'endpoint'
 node, according to the bindings defined in Documentation/devicetree/bindings/

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

@@ -10,6 +10,7 @@ Required properties:
    - "allwinner,sun6i-a31-i2s"
    - "allwinner,sun8i-a83t-i2s"
    - "allwinner,sun8i-h3-i2s"
+   - "allwinner,sun50i-a64-codec-i2s"
 - reg: physical base address of the controller and length of memory mapped
   region.
 - interrupts: should contain the I2S interrupt.
@@ -26,6 +27,7 @@ Required properties for the following compatibles:
 	- "allwinner,sun6i-a31-i2s"
 	- "allwinner,sun8i-a83t-i2s"
 	- "allwinner,sun8i-h3-i2s"
+	- "allwinner,sun50i-a64-codec-i2s"
 - resets: phandle to the reset line for this codec
 
 Example:

Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt

@@ -0,0 +1,12 @@
+* Allwinner A64 Codec Analog Controls
+
+Required properties:
+- compatible: must be one of the following compatibles:
+		- "allwinner,sun50i-a64-codec-analog"
+- reg: must contain the registers location and length
+
+Example:
+	codec_analog: codec-analog@1f015c0 {
+		compatible = "allwinner,sun50i-a64-codec-analog";
+		reg = <0x01f015c0 0x4>;
+	};

Documentation/devicetree/bindings/sound/ts3a227e.txt

@@ -14,7 +14,7 @@ Required properties:
 
 Optional properies:
  - ti,micbias:   Intended MICBIAS voltage (datasheet section 9.6.7).
-      Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage
+      Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage
       2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V
       Default value is "1" (2.2V).
 

Documentation/devicetree/bindings/sound/wm8782.txt

@@ -0,0 +1,17 @@
+WM8782 stereo ADC
+
+This device does not have any control interface or reset pins.
+
+Required properties:
+
+ - compatible  : "wlf,wm8782"
+ - Vdda-supply : phandle to a regulator for the analog power supply (2.7V - 5.5V)
+ - Vdd-supply  : phandle to a regulator for the digital power supply (2.7V - 3.6V)
+
+Example:
+
+wm8782: stereo-adc {
+	compatible = "wlf,wm8782";
+	Vdda-supply = <&vdda_supply>;
+	Vdd-supply = <&vdd_supply>;
+};

Documentation/devicetree/bindings/trivial-devices.txt

@@ -35,7 +35,6 @@ at,24c08		i2c serial eeprom  (24cxx)
 atmel,at97sc3204t	i2c trusted platform module (TPM)
 capella,cm32181		CM32181: Ambient Light Sensor
 capella,cm3232		CM3232: Ambient Light Sensor
-cirrus,cs42l51		Cirrus Logic CS42L51 audio codec
 dallas,ds1374		I2C, 32-Bit Binary Counter Watchdog RTC with Trickle Charger and Reset Input/Output
 dallas,ds1631		High-Precision Digital Thermometer
 dallas,ds1672		Dallas DS1672 Real-time Clock

Documentation/devicetree/bindings/vendor-prefixes.txt

@@ -235,6 +235,7 @@ micrel	Micrel Inc.
 microchip	Microchip Technology Inc.
 microcrystal	Micro Crystal AG
 micron	Micron Technology Inc.
+mikroe		MikroElektronika d.o.o.
 minix	MINIX Technology Ltd.
 miramems	MiraMEMS Sensing Technology Co., Ltd.
 mitsubishi	Mitsubishi Electric Corporation

MAINTAINERS

@@ -14619,6 +14619,13 @@ L:	netdev@vger.kernel.org
 S:	Maintained
 F:	drivers/net/ethernet/ti/netcp*
 
+TI PCM3060 ASoC CODEC DRIVER
+M:	Kirill Marinushkin <kmarinushkin@birdec.tech>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/pcm3060.txt
+F:	sound/soc/codecs/pcm3060*
+
 TI TAS571X FAMILY ASoC CODEC DRIVER
 M:	Kevin Cernekee <cernekee@chromium.org>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)

include/sound/simple_card_utils.h

@@ -51,29 +51,35 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
 
 #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, \
-				   dai_link->cpu_dai_name)
+				   dai_link->cpu_dai_name, NULL)
 #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,\
-				   dai_link->codec_dai_name)
+				   dai_link->codec_dai_name, dai_link->codecs)
 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,
-			       const char *name);
+			       const char *dai_name,
+			       struct snd_soc_dai_link_component *dlc);
 int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai);
 void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai);
 
 #define asoc_simple_card_parse_cpu(node, dai_link,				\
 				   list_name, cells_name, is_single_link)	\
-	asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node,		\
+	asoc_simple_card_parse_dai(node, NULL,					\
+		&dai_link->cpu_of_node,						\
 		&dai_link->cpu_dai_name, list_name, cells_name, is_single_link)
 #define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name)	\
-	asoc_simple_card_parse_dai(node, &dai_link->codec_of_node,		\
-		&dai_link->codec_dai_name, list_name, cells_name, NULL)
+	asoc_simple_card_parse_dai(node, dai_link->codecs,			\
+				   &dai_link->codec_of_node,			\
+				   &dai_link->codec_dai_name,			\
+				   list_name, cells_name, NULL)
 #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name)	\
-	asoc_simple_card_parse_dai(node, &dai_link->platform_of_node,		\
+	asoc_simple_card_parse_dai(node, dai_link->platform,					\
+		&dai_link->platform_of_node,					\
 		NULL, list_name, cells_name, NULL)
 int asoc_simple_card_parse_dai(struct device_node *node,
+				  struct snd_soc_dai_link_component *dlc,
 				  struct device_node **endpoint_np,
 				  const char **dai_name,
 				  const char *list_name,
@@ -81,12 +87,15 @@ int asoc_simple_card_parse_dai(struct device_node *node,
 				  int *is_single_links);
 
 #define asoc_simple_card_parse_graph_cpu(ep, dai_link)			\
-	asoc_simple_card_parse_graph_dai(ep, &dai_link->cpu_of_node,	\
+	asoc_simple_card_parse_graph_dai(ep, NULL,			\
+					 &dai_link->cpu_of_node,	\
 					 &dai_link->cpu_dai_name)
 #define asoc_simple_card_parse_graph_codec(ep, dai_link)		\
-	asoc_simple_card_parse_graph_dai(ep, &dai_link->codec_of_node,	\
+	asoc_simple_card_parse_graph_dai(ep, dai_link->codecs,		\
+					 &dai_link->codec_of_node,	\
 					 &dai_link->codec_dai_name)
 int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+				     struct snd_soc_dai_link_component *dlc,
 				     struct device_node **endpoint_np,
 				     const char **dai_name);
 

include/sound/soc-acpi-intel-match.h

@@ -25,4 +25,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[];
 
+/*
+ * generic table used for HDA codec-based platforms, possibly with
+ * additional ACPI-enumerated codecs
+ */
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[];
+
 #endif

include/sound/soc-dapm.h

@@ -406,12 +406,6 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 				 struct snd_soc_dai *dai);
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card);
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
-			 struct snd_soc_pcm_runtime *rtd,
-			 const struct snd_soc_pcm_stream *params,
-			 unsigned int num_params,
-			 struct snd_soc_dapm_widget *source,
-			 struct snd_soc_dapm_widget *sink);
 
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
@@ -590,9 +584,6 @@ struct snd_soc_dapm_widget {
 	void *priv;				/* widget specific data */
 	struct regulator *regulator;		/* attached regulator */
 	struct pinctrl *pinctrl;		/* attached pinctrl */
-	const struct snd_soc_pcm_stream *params; /* params for dai links */
-	unsigned int num_params; /* number of params for dai links */
-	unsigned int params_select; /* currently selected param for dai link */
 
 	/* dapm control */
 	int reg;				/* negative reg = no direct dapm */

include/sound/soc-dpcm.h

@@ -103,6 +103,16 @@ struct snd_soc_dpcm_runtime {
 	int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */
 };
 
+#define for_each_dpcm_fe(be, stream, dpcm)				\
+	list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe)
+
+#define for_each_dpcm_be(fe, stream, dpcm)				\
+	list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm)			\
+	list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+#define for_each_dpcm_be_rollback(fe, stream, dpcm)			\
+	list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+
 /* can this BE stop and free */
 int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
 		struct snd_soc_pcm_runtime *be, int stream);

include/sound/soc.h

@@ -372,6 +372,11 @@
 #define SND_SOC_COMP_ORDER_LATE		1
 #define SND_SOC_COMP_ORDER_LAST		2
 
+#define for_each_comp_order(order)		\
+	for (order  = SND_SOC_COMP_ORDER_FIRST;	\
+	     order <= SND_SOC_COMP_ORDER_LAST;	\
+	     order++)
+
 /*
  * Bias levels
  *
@@ -859,6 +864,11 @@ struct snd_soc_component {
 #endif
 };
 
+#define for_each_component_dais(component, dai)\
+	list_for_each_entry(dai, &(component)->dai_list, list)
+#define for_each_component_dais_safe(component, dai, _dai)\
+	list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list)
+
 struct snd_soc_rtdcom_list {
 	struct snd_soc_component *component;
 	struct list_head list; /* rtd::component_list */
@@ -915,6 +925,8 @@ struct snd_soc_dai_link {
 	 */
 	const char *platform_name;
 	struct device_node *platform_of_node;
+	struct snd_soc_dai_link_component *platform;
+
 	int id;	/* optional ID for machine driver link identification */
 
 	const struct snd_soc_pcm_stream *params;
@@ -976,6 +988,10 @@ struct snd_soc_dai_link {
 	struct list_head list; /* DAI link list of the soc card */
 	struct snd_soc_dobj dobj; /* For topology */
 };
+#define for_each_link_codecs(link, i, codec)				\
+	for ((i) = 0;							\
+	     ((i) < link->num_codecs) && ((codec) = &link->codecs[i]);	\
+	     (i)++)
 
 struct snd_soc_codec_conf {
 	/*
@@ -1054,7 +1070,6 @@ struct snd_soc_card {
 	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;
@@ -1092,6 +1107,7 @@ struct snd_soc_card {
 
 	/* lists of probed devices belonging to this card */
 	struct list_head component_dev_list;
+	struct list_head list;
 
 	struct list_head widgets;
 	struct list_head paths;
@@ -1114,6 +1130,23 @@ struct snd_soc_card {
 
 	void *drvdata;
 };
+#define for_each_card_prelinks(card, i, link)				\
+	for ((i) = 0;							\
+	     ((i) < (card)->num_links) && ((link) = &(card)->dai_link[i]); \
+	     (i)++)
+
+#define for_each_card_links(card, link)				\
+	list_for_each_entry(dai_link, &(card)->dai_link_list, list)
+#define for_each_card_links_safe(card, link, _link)			\
+	list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list)
+
+#define for_each_card_rtds(card, rtd)			\
+	list_for_each_entry(rtd, &(card)->rtd_list, list)
+#define for_each_card_rtds_safe(card, rtd, _rtd)	\
+	list_for_each_entry_safe(rtd, _rtd, &(card)->rtd_list, list)
+
+#define for_each_card_components(card, component)			\
+	list_for_each_entry(component, &(card)->component_dev_list, card_list)
 
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
 struct snd_soc_pcm_runtime {
@@ -1124,6 +1157,8 @@ struct snd_soc_pcm_runtime {
 	enum snd_soc_pcm_subclass pcm_subclass;
 	struct snd_pcm_ops ops;
 
+	unsigned int params_select; /* currently selected param for dai link */
+
 	/* Dynamic PCM BE runtime data */
 	struct snd_soc_dpcm_runtime dpcm[2];
 	int fe_compr;
@@ -1152,6 +1187,13 @@ struct snd_soc_pcm_runtime {
 	unsigned int dev_registered:1;
 	unsigned int pop_wait:1;
 };
+#define for_each_rtd_codec_dai(rtd, i, dai)\
+	for ((i) = 0;						       \
+	     ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
+	     (i)++)
+#define for_each_rtd_codec_dai_rollback(rtd, i, dai)		\
+	for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);)
+
 
 /* mixer control */
 struct soc_mixer_control {
@@ -1359,6 +1401,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
 	INIT_LIST_HEAD(&card->dapm_list);
 	INIT_LIST_HEAD(&card->aux_comp_list);
 	INIT_LIST_HEAD(&card->component_dev_list);
+	INIT_LIST_HEAD(&card->list);
 }
 
 static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)

sound/arm/Kconfig

@@ -31,7 +31,6 @@ endif	# SND_ARM
 
 config SND_PXA2XX_LIB
 	tristate
-	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
 	select SND_DMAENGINE_PCM
 
 config SND_PXA2XX_LIB_AC97

sound/pci/hda/hda_auto_parser.c

@@ -13,7 +13,7 @@
 #include <linux/export.h>
 #include <linux/sort.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 

sound/pci/hda/hda_beep.h

@@ -9,7 +9,7 @@
 #ifndef __SOUND_HDA_BEEP_H
 #define __SOUND_HDA_BEEP_H
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 
 #define HDA_BEEP_MODE_OFF	0
 #define HDA_BEEP_MODE_ON	1

sound/pci/hda/hda_bind.c

@@ -11,7 +11,7 @@
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 /*
@@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev)
 	hda_codec_patch_t patch;
 	int err;
 
+	if (codec->bus->core.ext_ops) {
+		if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach))
+			return -EINVAL;
+		return codec->bus->core.ext_ops->hdev_attach(&codec->core);
+	}
+
 	if (WARN_ON(!codec->preset))
 		return -EINVAL;
 
@@ -134,6 +140,12 @@ static int hda_codec_driver_remove(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
+	if (codec->bus->core.ext_ops) {
+		if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach))
+			return -EINVAL;
+		return codec->bus->core.ext_ops->hdev_detach(&codec->core);
+	}
+
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
 	snd_hda_codec_cleanup_for_unbind(codec);

sound/pci/hda/hda_codec.c

@@ -27,7 +27,7 @@
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include <sound/asoundef.h>
 #include <sound/tlv.h>
 #include <sound/initval.h>

sound/pci/hda/hda_controller.h

@@ -20,7 +20,7 @@
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include <sound/hda_register.h>
 
 #define AZX_MAX_CODECS		HDA_MAX_CODECS

sound/pci/hda/hda_eld.c

@@ -27,7 +27,7 @@
 #include <sound/core.h>
 #include <asm/unaligned.h>
 #include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 enum eld_versions {

sound/pci/hda/hda_generic.c

@@ -32,7 +32,7 @@
 #include <sound/core.h>
 #include <sound/jack.h>
 #include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/hda_hwdep.c

@@ -23,7 +23,7 @@
 #include <linux/compat.h>
 #include <linux/nospec.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>

sound/pci/hda/hda_intel.c

@@ -63,7 +63,7 @@
 #include <linux/vgaarb.h>
 #include <linux/vga_switcheroo.h>
 #include <linux/firmware.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_controller.h"
 #include "hda_intel.h"
 

sound/pci/hda/hda_jack.c

@@ -15,7 +15,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/hda_proc.c

@@ -25,7 +25,7 @@
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <linux/module.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 static int dump_coef = -1;

sound/pci/hda/hda_sysfs.c

@@ -14,7 +14,7 @@
 #include <linux/string.h>
 #include <linux/export.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>

sound/pci/hda/hda_tegra.c

@@ -35,7 +35,7 @@
 #include <sound/core.h>
 #include <sound/initval.h>
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_controller.h"
 
 /* Defines for Nvidia Tegra HDA support */

sound/pci/hda/patch_analog.c

@@ -24,7 +24,7 @@
 #include <linux/module.h>
 
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"

sound/pci/hda/patch_ca0110.c

@@ -22,7 +22,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/patch_ca0132.c

@@ -31,8 +31,9 @@
 #include <linux/types.h>
 #include <linux/io.h>
 #include <linux/pci.h>
+#include <asm/io.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/patch_cirrus.c

@@ -23,7 +23,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/patch_cmedia.c

@@ -25,7 +25,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/patch_conexant.c

@@ -27,7 +27,7 @@
 #include <sound/core.h>
 #include <sound/jack.h>
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"

sound/pci/hda/patch_hdmi.c

@@ -41,7 +41,7 @@
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
 #include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_jack.h"
 

sound/pci/hda/patch_realtek.c

@@ -32,7 +32,7 @@
 #include <linux/input.h>
 #include <sound/core.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/pci/hda/patch_si3054.c

@@ -27,7 +27,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 /* si3054 verbs */

sound/pci/hda/patch_sigmatel.c

@@ -32,7 +32,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"

sound/pci/hda/patch_via.c

@@ -52,7 +52,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"

sound/soc/amd/acp-da7219-max98357a.c

@@ -42,7 +42,7 @@
 #include "../codecs/da7219.h"
 #include "../codecs/da7219-aad.h"
 
-#define CZ_PLAT_CLK 25000000
+#define CZ_PLAT_CLK 48000000
 #define DUAL_CHANNEL		2
 
 static struct snd_soc_jack cz_jack;
@@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 	da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks");
 
 	ret = snd_soc_card_jack_new(card, "Headset Jack",
-				SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+				SND_JACK_HEADSET | SND_JACK_LINEOUT |
 				SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 				SND_JACK_BTN_2 | SND_JACK_BTN_3,
 				&cz_jack, NULL, 0);
@@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
 	.mask = 0,
 };
 
-static int cz_da7219_startup(struct snd_pcm_substream *substream)
+static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &constraints_rates);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->play_i2s_instance = I2S_SP_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL1;
 	return da7219_clk_enable(substream);
 }
@@ -162,11 +183,22 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_max_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->play_i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
@@ -177,21 +209,43 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
 static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL0;
 	return da7219_clk_enable(substream);
 }
@@ -201,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream)
 	da7219_clk_disable();
 }
 
+static const struct snd_soc_ops cz_da7219_play_ops = {
+	.startup = cz_da7219_play_startup,
+	.shutdown = cz_da7219_shutdown,
+};
+
 static const struct snd_soc_ops cz_da7219_cap_ops = {
-	.startup = cz_da7219_startup,
+	.startup = cz_da7219_cap_startup,
 	.shutdown = cz_da7219_shutdown,
 };
 
@@ -233,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
-		.ops = &cz_da7219_cap_ops,
+		.ops = &cz_da7219_play_ops,
 	},
 	{
 		.name = "amd-da7219-cap",

sound/soc/amd/acp-pcm-dma.c

@@ -867,8 +867,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 
 	if (pinfo) {
-		rtd->i2s_instance = pinfo->i2s_instance;
-		rtd->capture_channel = pinfo->capture_channel;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			rtd->i2s_instance = pinfo->play_i2s_instance;
+		} else {
+			rtd->i2s_instance = pinfo->cap_i2s_instance;
+			rtd->capture_channel = pinfo->capture_channel;
+		}
 	}
 	if (adata->asic_type == CHIP_STONEY) {
 		val = acp_reg_read(adata->acp_mmio,

sound/soc/amd/acp.h

@@ -158,7 +158,8 @@ struct audio_drv_data {
  * and dma driver
  */
 struct acp_platform_info {
-	u16 i2s_instance;
+	u16 play_i2s_instance;
+	u16 cap_i2s_instance;
 	u16 capture_channel;
 };
 

sound/soc/atmel/Kconfig

@@ -97,4 +97,16 @@ config SND_ATMEL_SOC_I2S
 	help
 	  Say Y or M if you want to add support for Atmel ASoc driver for boards
 	  using I2S.
+
+config SND_SOC_MIKROE_PROTO
+	tristate "Support for Mikroe-PROTO board"
+	depends on OF
+	depends on SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8731
+	help
+	  Say Y or M if you want to add support for MikroElektronika PROTO Audio
+	  Board. This board contains the WM8731 codec, which can be configured
+	  using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins.
+	  Both playback and capture are supported.
+
 endif

sound/soc/atmel/Makefile

@@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 snd-atmel-soc-classd-objs := atmel-classd.o
 snd-atmel-soc-pdmic-objs := atmel-pdmic.o
 snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o
+snd-soc-mikroe-proto-objs := mikroe-proto.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
@@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
 obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o
 obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o
+obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o

sound/soc/atmel/atmel_ssc_dai.c

@@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev)
 	struct ssc_device *ssc = dev_get_drvdata(dev);
 	int ret;
 
-	ret = snd_soc_register_component(dev, &atmel_ssc_component,
+	ret = devm_snd_soc_register_component(dev, &atmel_ssc_component,
 					 &atmel_ssc_dai, 1);
 	if (ret) {
 		dev_err(dev, "Could not register DAI: %d\n", ret);
-		goto err;
+		return ret;
 	}
 
 	if (ssc->pdata->use_dma)
@@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev)
 
 	if (ret) {
 		dev_err(dev, "Could not register PCM: %d\n", ret);
-		goto err_unregister_dai;
+		return ret;
 	}
 
 	return 0;
-
-err_unregister_dai:
-	snd_soc_unregister_component(dev);
-err:
-	return ret;
 }
 
 static void asoc_ssc_exit(struct device *dev)
@@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev)
 		atmel_pcm_dma_platform_unregister(dev);
 	else
 		atmel_pcm_pdc_platform_unregister(dev);
-
-	snd_soc_unregister_component(dev);
 }
 
 /**

sound/soc/atmel/mikroe-proto.c

@@ -0,0 +1,165 @@
+/*
+ * ASoC driver for PROTO AudioCODEC (with a WM8731)
+ *
+ * Author:      Florian Meier, <koalo@koalo.de>
+ *	      Copyright 2013
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8731.h"
+
+#define XTAL_RATE 12288000	/* This is fixed on this board */
+
+static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	/* Set proto sysclk */
+	int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+					 XTAL_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n",
+			ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget snd_proto_widget[] = {
+	SND_SOC_DAPM_MIC("Microphone Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route snd_proto_route[] = {
+	/* speaker connected to LHPOUT/RHPOUT */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Microphone Jack"},
+};
+
+/* audio machine driver */
+static struct snd_soc_card snd_proto = {
+	.name		= "snd_mikroe_proto",
+	.owner		= THIS_MODULE,
+	.dapm_widgets	= snd_proto_widget,
+	.num_dapm_widgets = ARRAY_SIZE(snd_proto_widget),
+	.dapm_routes	= snd_proto_route,
+	.num_dapm_routes = ARRAY_SIZE(snd_proto_route),
+};
+
+static int snd_proto_probe(struct platform_device *pdev)
+{
+	struct snd_soc_dai_link *dai;
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	unsigned int dai_fmt;
+	int ret = 0;
+
+	if (!np) {
+		dev_err(&pdev->dev, "No device node supplied\n");
+		return -EINVAL;
+	}
+
+	snd_proto.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&snd_proto, "model");
+	if (ret)
+		return ret;
+
+	dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
+	snd_proto.dai_link = dai;
+	snd_proto.num_links = 1;
+
+	dai->name = "WM8731";
+	dai->stream_name = "WM8731 HiFi";
+	dai->codec_dai_name = "wm8731-hifi";
+	dai->init = &snd_proto_init;
+
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "audio-codec node missing\n");
+		return -EINVAL;
+	}
+	dai->codec_of_node = codec_np;
+
+	cpu_np = of_parse_phandle(np, "i2s-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "i2s-controller missing\n");
+		return -EINVAL;
+	}
+	dai->cpu_of_node = cpu_np;
+	dai->platform_of_node = cpu_np;
+
+	dai_fmt = snd_soc_of_parse_daifmt(np, NULL,
+					  &bitclkmaster, &framemaster);
+	if (bitclkmaster != framemaster) {
+		dev_err(&pdev->dev, "Must be the same bitclock and frame master\n");
+		return -EINVAL;
+	}
+	if (bitclkmaster) {
+		dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+		if (codec_np == bitclkmaster)
+			dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+		else
+			dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	}
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+	dai->dai_fmt = dai_fmt;
+
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	ret = snd_soc_register_card(&snd_proto);
+	if (ret && ret != -EPROBE_DEFER)
+		dev_err(&pdev->dev,
+			"snd_soc_register_card() failed: %d\n", ret);
+
+	return ret;
+}
+
+static int snd_proto_remove(struct platform_device *pdev)
+{
+	return snd_soc_unregister_card(&snd_proto);
+}
+
+static const struct of_device_id snd_proto_of_match[] = {
+	{ .compatible = "mikroe,mikroe-proto", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, snd_proto_of_match);
+
+static struct platform_driver snd_proto_driver = {
+	.driver = {
+		.name   = "snd-mikroe-proto",
+		.of_match_table = snd_proto_of_match,
+	},
+	.probe	  = snd_proto_probe,
+	.remove	 = snd_proto_remove,
+};
+
+module_platform_driver(snd_proto_driver);
+
+MODULE_AUTHOR("Florian Meier");
+MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)");
+MODULE_LICENSE("GPL");

sound/soc/atmel/tse850-pcm5142.c

@@ -1,44 +1,38 @@
-/*
- * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
- *
- * Copyright (C) 2016 Axentia Technologies AB
- *
- * Author: Peter Rosin <peda@axentia.se>
- *
- * 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.
- */
-
-/*
- *               loop1 relays
- *   IN1 +---o  +------------+  o---+ OUT1
- *            \                /
- *             +              +
- *             |   /          |
- *             +--o  +--.     |
- *             |  add   |     |
- *             |        V     |
- *             |      .---.   |
- *   DAC +----------->|Sum|---+
- *             |      '---'   |
- *             |              |
- *             +              +
- *
- *   IN2 +---o--+------------+--o---+ OUT2
- *               loop2 relays
- *
- * The 'loop1' gpio pin controlls two relays, which are either in loop
- * position, meaning that input and output are directly connected, or
- * they are in mixer position, meaning that the signal is passed through
- * the 'Sum' mixer. Similarly for 'loop2'.
- *
- * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
- * mixer (if 'add' is active) and feeding the mixer output to OUT1. The
- * 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
- * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
- * of the (filtered) output from the PCM5142 codec.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
+//
+// Copyright (C) 2016 Axentia Technologies AB
+//
+// Author: Peter Rosin <peda@axentia.se>
+//
+//               loop1 relays
+//   IN1 +---o  +------------+  o---+ OUT1
+//            \                /
+//             +              +
+//             |   /          |
+//             +--o  +--.     |
+//             |  add   |     |
+//             |        V     |
+//             |      .---.   |
+//   DAC +----------->|Sum|---+
+//             |      '---'   |
+//             |              |
+//             +              +
+//
+//   IN2 +---o--+------------+--o---+ OUT2
+//               loop2 relays
+//
+// The 'loop1' gpio pin controlls two relays, which are either in loop
+// position, meaning that input and output are directly connected, or
+// they are in mixer position, meaning that the signal is passed through
+// the 'Sum' mixer. Similarly for 'loop2'.
+//
+// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
+// mixer (if 'add' is active) and feeding the mixer output to OUT1. The
+// 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
+// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
+// of the (filtered) output from the PCM5142 codec.
 
 #include <linux/clk.h>
 #include <linux/gpio.h>
@@ -452,4 +446,4 @@ module_platform_driver(tse850_driver);
 /* Module information */
 MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
 MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");

sound/soc/bcm/cygnus-ssp.c

@@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	cygaud->active_ports = 0;
 
 	dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
-	err = snd_soc_register_component(dev, &cygnus_ssp_component,
+	err = devm_snd_soc_register_component(dev, &cygnus_ssp_component,
 				cygnus_ssp_dai, active_port_count);
 	if (err) {
 		dev_err(dev, "snd_soc_register_dai failed\n");
@@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	if (cygaud->irq_num <= 0) {
 		dev_err(dev, "platform_get_irq failed\n");
 		err = cygaud->irq_num;
-		goto err_irq;
+		return err;
 	}
 
 	err = audio_clk_init(pdev, cygaud);
 	if (err) {
 		dev_err(dev, "audio clock initialization failed\n");
-		goto err_irq;
+		return err;
 	}
 
 	err = cygnus_soc_platform_register(dev, cygaud);
 	if (err) {
 		dev_err(dev, "platform reg error %d\n", err);
-		goto err_irq;
+		return err;
 	}
 
 	return 0;
-
-err_irq:
-	snd_soc_unregister_component(dev);
-	return err;
 }
 
 static int cygnus_ssp_remove(struct platform_device *pdev)
 {
 	cygnus_soc_platform_unregister(&pdev->dev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }

sound/soc/codecs/Kconfig

@@ -82,6 +82,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ES7241
 	select SND_SOC_GTM601
 	select SND_SOC_HDAC_HDMI
+	select SND_SOC_HDAC_HDA
 	select SND_SOC_ICS43432
 	select SND_SOC_INNO_RK3036
 	select SND_SOC_ISABELLE if I2C
@@ -109,6 +110,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MT6351 if MTK_PMIC_WRAP
 	select SND_SOC_NAU8540 if I2C
 	select SND_SOC_NAU8810 if I2C
+	select SND_SOC_NAU8822 if I2C
 	select SND_SOC_NAU8824 if I2C
 	select SND_SOC_NAU8825 if I2C
 	select SND_SOC_HDMI_CODEC
@@ -119,6 +121,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM186X_I2C if I2C
 	select SND_SOC_PCM186X_SPI if SPI_MASTER
 	select SND_SOC_PCM3008
+	select SND_SOC_PCM3060_I2C if I2C
+	select SND_SOC_PCM3060_SPI if SPI_MASTER
 	select SND_SOC_PCM3168A_I2C if I2C
 	select SND_SOC_PCM3168A_SPI if SPI_MASTER
 	select SND_SOC_PCM5102A
@@ -575,7 +579,11 @@ config SND_SOC_DA9055
 	tristate
 
 config SND_SOC_DMIC
-	tristate
+	tristate "Generic Digital Microphone CODEC"
+	depends on GPIOLIB
+	help
+	  Enable support for the Generic Digital Microphone CODEC.
+	  Select this if your sound card has DMICs.
 
 config SND_SOC_HDMI_CODEC
 	tristate
@@ -615,6 +623,10 @@ config SND_SOC_HDAC_HDMI
 	select SND_PCM_ELD
 	select HDMI
 
+config SND_SOC_HDAC_HDA
+	tristate
+	select SND_HDA
+
 config SND_SOC_ICS43432
 	tristate
 
@@ -629,7 +641,8 @@ config SND_SOC_LM49453
 	tristate
 
 config SND_SOC_MAX98088
-       tristate
+	tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
+	depends on I2C
 
 config SND_SOC_MAX98090
        tristate
@@ -732,6 +745,21 @@ config SND_SOC_PCM186X_SPI
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_PCM3060
+       tristate
+
+config SND_SOC_PCM3060_I2C
+	tristate "Texas Instruments PCM3060 CODEC - I2C"
+	depends on I2C
+	select SND_SOC_PCM3060
+	select REGMAP_I2C
+
+config SND_SOC_PCM3060_SPI
+	tristate "Texas Instruments PCM3060 CODEC - SPI"
+	depends on SPI_MASTER
+	select SND_SOC_PCM3060
+	select REGMAP_SPI
+
 config SND_SOC_PCM3168A
 	tristate
 
@@ -1299,6 +1327,10 @@ config SND_SOC_NAU8810
 	tristate "Nuvoton Technology Corporation NAU88C10 CODEC"
 	depends on I2C
 
+config SND_SOC_NAU8822
+	tristate "Nuvoton Technology Corporation NAU88C22 CODEC"
+	depends on I2C
+
 config SND_SOC_NAU8824
 	tristate "Nuvoton Technology Corporation NAU88L24 CODEC"
 	depends on I2C

sound/soc/codecs/Makefile

@@ -78,6 +78,7 @@ 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-hdac-hda-objs := hdac_hda.o
 snd-soc-ics43432-objs := ics43432.o
 snd-soc-inno-rk3036-objs := inno_rk3036.o
 snd-soc-isabelle-objs := isabelle.o
@@ -106,6 +107,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
 snd-soc-mt6351-objs := mt6351.o
 snd-soc-nau8540-objs := nau8540.o
 snd-soc-nau8810-objs := nau8810.o
+snd-soc-nau8822-objs := nau8822.o
 snd-soc-nau8824-objs := nau8824.o
 snd-soc-nau8825-objs := nau8825.o
 snd-soc-hdmi-codec-objs := hdmi-codec.o
@@ -119,6 +121,9 @@ snd-soc-pcm186x-objs := pcm186x.o
 snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o
 snd-soc-pcm186x-spi-objs := pcm186x-spi.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm3060-objs := pcm3060.o
+snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o
+snd-soc-pcm3060-spi-objs := pcm3060-spi.o
 snd-soc-pcm3168a-objs := pcm3168a.o
 snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
 snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
@@ -338,6 +343,7 @@ 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-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o
 obj-$(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
@@ -366,6 +372,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o
 obj-$(CONFIG_SND_SOC_MT6351)	+= snd-soc-mt6351.o
 obj-$(CONFIG_SND_SOC_NAU8540)   += snd-soc-nau8540.o
 obj-$(CONFIG_SND_SOC_NAU8810)   += snd-soc-nau8810.o
+obj-$(CONFIG_SND_SOC_NAU8822)   += snd-soc-nau8822.o
 obj-$(CONFIG_SND_SOC_NAU8824)   += snd-soc-nau8824.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_HDMI_CODEC)	+= snd-soc-hdmi-codec.o
@@ -379,6 +386,9 @@ obj-$(CONFIG_SND_SOC_PCM186X)	+= snd-soc-pcm186x.o
 obj-$(CONFIG_SND_SOC_PCM186X_I2C)	+= snd-soc-pcm186x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM186X_SPI)	+= snd-soc-pcm186x-spi.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM3060)	+= snd-soc-pcm3060.o
+obj-$(CONFIG_SND_SOC_PCM3060_I2C)	+= snd-soc-pcm3060-i2c.o
+obj-$(CONFIG_SND_SOC_PCM3060_SPI)	+= snd-soc-pcm3060-spi.o
 obj-$(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

sound/soc/codecs/adau1761.c

@@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component)
 			ARRAY_SIZE(adau1761_jack_detect_controls));
 		if (ret)
 			return ret;
-	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
+		/* fall through */
+	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE:
 		ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
 			ARRAY_SIZE(adau1761_no_dmic_routes));
 		if (ret)

sound/soc/codecs/adau17x1.c

@@ -21,11 +21,18 @@
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
 #include <linux/regmap.h>
+#include <asm/unaligned.h>
 
 #include "sigmadsp.h"
 #include "adau17x1.h"
 #include "adau-utils.h"
 
+#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006
+#define ADAU17X1_SAFELOAD_TRIGGER 0x0007
+#define ADAU17X1_SAFELOAD_DATA 0x0001
+#define ADAU17X1_SAFELOAD_DATA_SIZE 20
+#define ADAU17X1_WORD_SIZE 4
+
 static const char * const adau17x1_capture_mixer_boost_text[] = {
 	"Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
 };
@@ -60,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = {
 	SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum),
 };
 
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate);
+
 static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -313,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
 	{ "Capture", NULL, "Right Decimator" },
 };
 
-bool adau17x1_has_dsp(struct adau *adau)
+static bool adau17x1_has_dsp(struct adau *adau)
 {
 	switch (adau->type) {
 	case ADAU1761:
@@ -324,7 +334,17 @@ bool adau17x1_has_dsp(struct adau *adau)
 		return false;
 	}
 }
-EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
+
+static bool adau17x1_has_safeload(struct adau *adau)
+{
+	switch (adau->type) {
+	case ADAU1761:
+	case ADAU1781:
+		return true;
+	default:
+		return false;
+	}
+}
 
 static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
 	int source, unsigned int freq_in, unsigned int freq_out)
@@ -836,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
 	unsigned int rate)
 {
 	int ret;
@@ -880,7 +900,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component,
 
 	return ret;
 }
-EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
 int adau17x1_add_widgets(struct snd_soc_component *component)
 {
@@ -957,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component)
 }
 EXPORT_SYMBOL_GPL(adau17x1_resume);
 
+static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr,
+	const uint8_t bytes[], size_t len)
+{
+	uint8_t buf[ADAU17X1_WORD_SIZE];
+	uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE];
+	unsigned int addr_offset;
+	unsigned int nbr_words;
+	int ret;
+
+	/* write data to safeload addresses. Check if len is not a multiple of
+	 * 4 bytes, if so we need to zero pad.
+	 */
+	nbr_words = len / ADAU17X1_WORD_SIZE;
+	if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) {
+		ret = regmap_raw_write(sigmadsp->control_data,
+			ADAU17X1_SAFELOAD_DATA, bytes, len);
+	} else {
+		nbr_words++;
+		memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE);
+		memcpy(data, bytes, len);
+		ret = regmap_raw_write(sigmadsp->control_data,
+			ADAU17X1_SAFELOAD_DATA, data,
+			nbr_words * ADAU17X1_WORD_SIZE);
+	}
+
+	if (ret < 0)
+		return ret;
+
+	/* Write target address, target address is offset by 1 */
+	addr_offset = addr - 1;
+	put_unaligned_be32(addr_offset, buf);
+	ret = regmap_raw_write(sigmadsp->control_data,
+		ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE);
+	if (ret < 0)
+		return ret;
+
+	/* write nbr of words to trigger address */
+	put_unaligned_be32(nbr_words, buf);
+	ret = regmap_raw_write(sigmadsp->control_data,
+		ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static const struct sigmadsp_ops adau17x1_sigmadsp_ops = {
+	.safeload = adau17x1_safeload,
+};
+
 int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	enum adau17x1_type type, void (*switch_mode)(struct device *dev),
 	const char *firmware_name)
@@ -1002,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	dev_set_drvdata(dev, adau);
 
 	if (firmware_name) {
-		adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL,
-			firmware_name);
+		if (adau17x1_has_safeload(adau)) {
+			adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+				&adau17x1_sigmadsp_ops, firmware_name);
+		} else {
+			adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+				NULL, firmware_name);
+		}
 		if (IS_ERR(adau->sigmadsp)) {
 			dev_warn(dev, "Could not find firmware file: %ld\n",
 				PTR_ERR(adau->sigmadsp));

sound/soc/codecs/adau17x1.h

@@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
-	unsigned int rate);
-bool adau17x1_has_dsp(struct adau *adau);
-
 #define ADAU17X1_CLOCK_CONTROL			0x4000
 #define ADAU17X1_PLL_CONTROL			0x4002
 #define ADAU17X1_REC_POWER_MGMT			0x4009

sound/soc/codecs/cs4265.c

@@ -154,11 +154,11 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = {
 	SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
 				6, 1, 0),
 	SOC_ENUM("C Data Access", cam_mode_enum),
+	SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1),
 	SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
 				3, 1, 0),
 	SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
-	SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2,
-				0, 1, 0),
+	SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0),
 	SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
 	SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
 };
@@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = {
 	{"LINEOUTR", NULL, "DAC"},
 	{"SPDIFOUT", NULL, "SPDIF"},
 
+	{"Pre-amp MIC", NULL, "MICL"},
+	{"Pre-amp MIC", NULL, "MICR"},
+	{"ADC Mux", "MIC", "Pre-amp MIC"},
 	{"ADC Mux", "LINEIN", "LINEINL"},
 	{"ADC Mux", "LINEIN", "LINEINR"},
-	{"ADC Mux", "MIC", "MICL"},
-	{"ADC Mux", "MIC", "MICR"},
 	{"ADC", NULL, "ADC Mux"},
 	{"DOUT", NULL, "ADC"},
 	{"DAI1 Capture", NULL, "DOUT"},
@@ -496,7 +497,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component,
 			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
 
 #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
-			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
 
 static const struct snd_soc_dai_ops cs4265_ops = {
 	.hw_params	= cs4265_pcm_hw_params,

sound/soc/codecs/cs42l51.c

@@ -21,6 +21,7 @@
  *  - master mode *NOT* supported
  */
 
+#include <linux/clk.h>
 #include <linux/module.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -41,6 +42,7 @@ enum master_slave_mode {
 
 struct cs42l51_private {
 	unsigned int mclk;
+	struct clk *mclk_handle;
 	unsigned int audio_mode;	/* The mode (I2S or left-justified) */
 	enum master_slave_mode func;
 };
@@ -237,6 +239,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
 		&cs42l51_adcr_mux_controls),
 };
 
+static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
+	SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+};
+
 static const struct snd_soc_dapm_route cs42l51_routes[] = {
 	{"HPL", NULL, "Left DAC"},
 	{"HPR", NULL, "Right DAC"},
@@ -487,6 +493,14 @@ static struct snd_soc_dai_driver cs42l51_dai = {
 static int cs42l51_component_probe(struct snd_soc_component *component)
 {
 	int ret, reg;
+	struct snd_soc_dapm_context *dapm;
+	struct cs42l51_private *cs42l51;
+
+	cs42l51 = snd_soc_component_get_drvdata(component);
+	dapm = snd_soc_component_get_dapm(component);
+
+	if (cs42l51->mclk_handle)
+		snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1);
 
 	/*
 	 * DAC configuration
@@ -540,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap)
 
 	dev_set_drvdata(dev, cs42l51);
 
+	cs42l51->mclk_handle = devm_clk_get(dev, "MCLK");
+	if (IS_ERR(cs42l51->mclk_handle)) {
+		if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT)
+			return PTR_ERR(cs42l51->mclk_handle);
+		cs42l51->mclk_handle = NULL;
+	}
+
 	/* Verify that we have a CS42L51 */
 	ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
 	if (ret < 0) {

sound/soc/codecs/dmic.c

@@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = {
 	{.compatible = "dmic-codec"},
 	{}
 };
+MODULE_DEVICE_TABLE(of, dmic_dev_match);
 
 static struct platform_driver dmic_driver = {
 	.driver = {

sound/soc/codecs/es8328.c

@@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
 		break;
 	case 22579200:
 		mclkdiv2 = 1;
-		/* fallthru */
+		/* fall through */
 	case 11289600:
 		es8328->sysclk_constraints = &constraints_11289;
 		es8328->mclk_ratios = ratios_11289;
 		break;
 	case 24576000:
 		mclkdiv2 = 1;
-		/* fallthru */
+		/* fall through */
 	case 12288000:
 		es8328->sysclk_constraints = &constraints_12288;
 		es8328->mclk_ratios = ratios_12288;

sound/soc/codecs/hdac_hda.c

@@ -0,0 +1,483 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers
+ * with ASoC platform drivers. These APIs are called by the legacy HDA
+ * codec drivers using hdac_ext_bus_ops ops.
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
+#include <sound/hda_register.h>
+#include "hdac_hda.h"
+
+#define HDAC_ANALOG_DAI_ID		0
+#define HDAC_DIGITAL_DAI_ID		1
+#define HDAC_ALT_ANALOG_DAI_ID		2
+
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+			SNDRV_PCM_FMTBIT_U8 | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | \
+			SNDRV_PCM_FMTBIT_U32_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai);
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai);
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai);
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai);
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				     unsigned int tx_mask, unsigned int rx_mask,
+				     int slots, int slot_width);
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+						 struct snd_soc_dai *dai);
+
+static struct snd_soc_dai_ops hdac_hda_dai_ops = {
+	.startup = hdac_hda_dai_open,
+	.shutdown = hdac_hda_dai_close,
+	.prepare = hdac_hda_dai_prepare,
+	.hw_free = hdac_hda_dai_hw_free,
+	.set_tdm_slot = hdac_hda_dai_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver hdac_hda_dais[] = {
+{
+	.id = HDAC_ANALOG_DAI_ID,
+	.name = "Analog Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name	= "Analog Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 16,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= STUB_FORMATS,
+		.sig_bits	= 24,
+	},
+	.capture = {
+		.stream_name    = "Analog Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+},
+{
+	.id = HDAC_DIGITAL_DAI_ID,
+	.name = "Digital Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name    = "Digital Codec Playback",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates          = SNDRV_PCM_RATE_8000_192000,
+		.formats        = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+	.capture = {
+		.stream_name    = "Digital Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+},
+{
+	.id = HDAC_ALT_ANALOG_DAI_ID,
+	.name = "Alt Analog Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name	= "Alt Analog Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 16,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= STUB_FORMATS,
+		.sig_bits	= 24,
+	},
+	.capture = {
+		.stream_name    = "Alt Analog Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+}
+
+};
+
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				     unsigned int tx_mask, unsigned int rx_mask,
+				     int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hdac_hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = &hda_pvt->pcm[dai->id];
+	if (tx_mask)
+		pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+	else
+		pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+
+	return 0;
+}
+
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	hda_stream = &pcm->stream[substream->stream];
+	snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream);
+
+	return 0;
+}
+
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hdac_device *hdev;
+	struct hda_pcm_stream *hda_stream;
+	unsigned int format_val;
+	struct hda_pcm *pcm;
+	unsigned int stream;
+	int ret = 0;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	hdev = &hda_pvt->codec.core;
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	format_val = snd_hdac_calc_stream_format(runtime->rate,
+						 runtime->channels,
+						 runtime->format,
+						 hda_stream->maxbps,
+						 0);
+	if (!format_val) {
+		dev_err(&hdev->dev,
+			"invalid format_val, rate=%d, ch=%d, format=%d\n",
+			runtime->rate, runtime->channels, runtime->format);
+		return -EINVAL;
+	}
+
+	stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream];
+
+	ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream,
+				    stream, format_val, substream);
+	if (ret < 0)
+		dev_err(&hdev->dev, "codec prepare failed %d\n", ret);
+
+	return ret;
+}
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+	int ret;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	snd_hda_codec_pcm_get(pcm);
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream);
+	if (ret < 0)
+		snd_hda_codec_pcm_put(pcm);
+
+	return ret;
+}
+
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return;
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream);
+
+	snd_hda_codec_pcm_put(pcm);
+}
+
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+						 struct snd_soc_dai *dai)
+{
+	struct hda_codec *hcodec = &hda_pvt->codec;
+	struct hda_pcm *cpcm;
+	const char *pcm_name;
+
+	switch (dai->id) {
+	case HDAC_ANALOG_DAI_ID:
+		pcm_name = "Analog";
+		break;
+	case HDAC_DIGITAL_DAI_ID:
+		pcm_name = "Digital";
+		break;
+	case HDAC_ALT_ANALOG_DAI_ID:
+		pcm_name = "Alt Analog";
+		break;
+	default:
+		dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id);
+		return NULL;
+	}
+
+	list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) {
+		if (strpbrk(cpcm->name, pcm_name))
+			return cpcm;
+	}
+
+	dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name);
+	return NULL;
+}
+
+static int hdac_hda_codec_probe(struct snd_soc_component *component)
+{
+	struct hdac_hda_priv *hda_pvt =
+			snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
+	struct hdac_device *hdev = &hda_pvt->codec.core;
+	struct hda_codec *hcodec = &hda_pvt->codec;
+	struct hdac_ext_link *hlink;
+	hda_codec_patch_t patch;
+	int ret;
+
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+	ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card,
+				       hdev->addr, hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "failed to create hda codec %d\n", ret);
+		goto error_no_pm;
+	}
+
+	/*
+	 * snd_hda_codec_device_new decrements the usage count so call get pm
+	 * else the device will be powered off
+	 */
+	pm_runtime_get_noresume(&hdev->dev);
+
+	hcodec->bus->card = dapm->card->snd_card;
+
+	ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
+		goto error;
+	}
+
+	ret = snd_hdac_regmap_init(&hcodec->core);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "regmap init failed\n");
+		goto error;
+	}
+
+	patch = (hda_codec_patch_t)hcodec->preset->driver_data;
+	if (patch) {
+		ret = patch(hcodec);
+		if (ret < 0) {
+			dev_err(&hdev->dev, "patch failed %d\n", ret);
+			goto error;
+		}
+	} else {
+		dev_dbg(&hdev->dev, "no patch file found\n");
+	}
+
+	ret = snd_hda_codec_parse_pcms(hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
+		goto error;
+	}
+
+	ret = snd_hda_codec_build_controls(hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "unable to create controls %d\n", ret);
+		goto error;
+	}
+
+	hcodec->core.lazy_cache = true;
+
+	/*
+	 * hdac_device core already sets the state to active and calls
+	 * get_noresume. So enable runtime and set the device to suspend.
+	 * pm_runtime_enable is also called during codec registeration
+	 */
+	pm_runtime_put(&hdev->dev);
+	pm_runtime_suspend(&hdev->dev);
+
+	return 0;
+
+error:
+	pm_runtime_put(&hdev->dev);
+error_no_pm:
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+	return ret;
+}
+
+static void hdac_hda_codec_remove(struct snd_soc_component *component)
+{
+	struct hdac_hda_priv *hda_pvt =
+		      snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = &hda_pvt->codec.core;
+	struct hdac_ext_link *hlink = NULL;
+
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return;
+	}
+
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+	pm_runtime_disable(&hdev->dev);
+}
+
+static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = {
+	{"AIF1TX", NULL, "Codec Input Pin1"},
+	{"AIF2TX", NULL, "Codec Input Pin2"},
+	{"AIF3TX", NULL, "Codec Input Pin3"},
+
+	{"Codec Output Pin1", NULL, "AIF1RX"},
+	{"Codec Output Pin2", NULL, "AIF2RX"},
+	{"Codec Output Pin3", NULL, "AIF3RX"},
+};
+
+static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = {
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+
+	/* Input Pins */
+	SND_SOC_DAPM_INPUT("Codec Input Pin1"),
+	SND_SOC_DAPM_INPUT("Codec Input Pin2"),
+	SND_SOC_DAPM_INPUT("Codec Input Pin3"),
+
+	/* Output Pins */
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin1"),
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin2"),
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin3"),
+};
+
+static const struct snd_soc_component_driver hdac_hda_codec = {
+	.probe		= hdac_hda_codec_probe,
+	.remove		= hdac_hda_codec_remove,
+	.idle_bias_on	= false,
+	.dapm_widgets           = hdac_hda_dapm_widgets,
+	.num_dapm_widgets       = ARRAY_SIZE(hdac_hda_dapm_widgets),
+	.dapm_routes            = hdac_hda_dapm_routes,
+	.num_dapm_routes        = ARRAY_SIZE(hdac_hda_dapm_routes),
+};
+
+static int hdac_hda_dev_probe(struct hdac_device *hdev)
+{
+	struct hdac_ext_link *hlink;
+	struct hdac_hda_priv *hda_pvt;
+	int ret;
+
+	/* hold the ref while we probe */
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return -EIO;
+	}
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+	hda_pvt = hdac_to_hda_priv(hdev);
+	if (!hda_pvt)
+		return -ENOMEM;
+
+	/* ASoC specific initialization */
+	ret = devm_snd_soc_register_component(&hdev->dev,
+					 &hdac_hda_codec, hdac_hda_dais,
+					 ARRAY_SIZE(hdac_hda_dais));
+	if (ret < 0) {
+		dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
+		return ret;
+	}
+
+	dev_set_drvdata(&hdev->dev, hda_pvt);
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+
+	return ret;
+}
+
+static int hdac_hda_dev_remove(struct hdac_device *hdev)
+{
+	return 0;
+}
+
+static struct hdac_ext_bus_ops hdac_ops = {
+	.hdev_attach = hdac_hda_dev_probe,
+	.hdev_detach = hdac_hda_dev_remove,
+};
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void)
+{
+	return &hdac_ops;
+}
+EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers");
+MODULE_AUTHOR("Rakesh Ughreja<rakesh.a.ughreja@intel.com>");

sound/soc/codecs/hdac_hda.h

@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+#ifndef __HDAC_HDA_H__
+#define __HDAC_HDA_H__
+
+struct hdac_hda_pcm {
+	int stream_tag[2];
+};
+
+struct hdac_hda_priv {
+	struct hda_codec codec;
+	struct hdac_hda_pcm pcm[2];
+};
+
+#define hdac_to_hda_priv(_hdac) \
+			container_of(_hdac, struct hdac_hda_priv, codec.core)
+#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void);
+
+#endif /* __HDAC_HDA_H__ */

sound/soc/codecs/hdac_hdmi.c

@@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev,
 		if (ret)
 			return ret;
 
+		/* Filter out 44.1, 88.2 and 176.4Khz */
+		rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+			   SNDRV_PCM_RATE_176400);
+		if (!rates)
+			return -EINVAL;
+
 		sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
 		hdmi_dais[i].name = devm_kstrdup(&hdev->dev,
 					dai_name, GFP_KERNEL);
@@ -1598,7 +1604,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->pcm && (rtd->pcm->device == device))
 			return rtd->pcm;
 	}
@@ -1961,9 +1967,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx)
 
 	port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head);
 
-	if (!port)
-		return 0;
-
 	if (!port || !port->eld.eld_valid)
 		return 0;
 

sound/soc/codecs/max98088.c

@@ -16,6 +16,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
+#include <linux/clk.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -42,6 +43,7 @@ struct max98088_priv {
 	struct regmap *regmap;
 	enum max98088_type devtype;
 	struct max98088_pdata *pdata;
+	struct clk *mclk;
 	unsigned int sysclk;
 	struct max98088_cdata dai[2];
 	int eq_textcnt;
@@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai,
        if (freq == max98088->sysclk)
                return 0;
 
+	if (!IS_ERR(max98088->mclk)) {
+		freq = clk_round_rate(max98088->mclk, freq);
+		clk_set_rate(max98088->mclk, freq);
+	}
+
        /* Setup clocks for slave mode, and using the PLL
         * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
         *         0x02 (when master clk is 20MHz to 30MHz)..
@@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
+		/*
+		 * SND_SOC_BIAS_PREPARE is called while preparing for a
+		 * transition to ON or away from ON. If current bias_level
+		 * is SND_SOC_BIAS_ON, then it is preparing for a transition
+		 * away from ON. Disable the clock in that case, otherwise
+		 * enable it.
+		 */
+		if (!IS_ERR(max98088->mclk)) {
+			if (snd_soc_component_get_bias_level(component) ==
+			    SND_SOC_BIAS_ON)
+				clk_disable_unprepare(max98088->mclk);
+			else
+				clk_prepare_enable(max98088->mclk);
+		}
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        if (IS_ERR(max98088->regmap))
 	       return PTR_ERR(max98088->regmap);
 
+	max98088->mclk = devm_clk_get(&i2c->dev, "mclk");
+	if (IS_ERR(max98088->mclk))
+		if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER)
+			return PTR_ERR(max98088->mclk);
+
        max98088->devtype = id->driver_data;
 
        i2c_set_clientdata(i2c, max98088);
@@ -1742,9 +1768,19 @@ static const struct i2c_device_id max98088_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id max98088_of_match[] = {
+	{ .compatible = "maxim,max98088" },
+	{ .compatible = "maxim,max98089" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, max98088_of_match);
+#endif
+
 static struct i2c_driver max98088_i2c_driver = {
 	.driver = {
 		.name = "max98088",
+		.of_match_table = of_match_ptr(max98088_of_match),
 	},
 	.probe  = max98088_i2c_probe,
 	.id_table = max98088_i2c_id,

sound/soc/codecs/max98373.c

@@ -2,6 +2,7 @@
 // Copyright (c) 2017, Maxim Integrated
 
 #include <linux/acpi.h>
+#include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/module.h>
 #include <linux/regmap.h>
@@ -454,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"),
 SND_SOC_DAPM_SIGGEN("FBMON"),
 };
 
-static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0);
+static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1);
 static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv,
 	0, 8, TLV_DB_SCALE_ITEM(0, 50, 0),
 	9, 10, TLV_DB_SCALE_ITEM(500, 100, 0),
@@ -470,19 +471,19 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv,
 	0, 9, TLV_DB_SCALE_ITEM(800, 100, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv,
-	0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
-	2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0),
-	8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0),
-	10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0),
-	12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
-	14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+	0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0),
+	2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0),
+	5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+	7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0),
+	10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0),
+	14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv,
-	0, 15, TLV_DB_SCALE_ITEM(0, -100, 0),
+	0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0),
 );
 
 static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
-	0, 60, TLV_DB_SCALE_ITEM(0, -25, 0),
+	0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0),
 );
 
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
@@ -604,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG,
 SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG,
 	MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0),
 SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL,
-	0, 0x7F, 0, max98373_digital_tlv),
+	0, 0x7F, 1, max98373_digital_tlv),
 SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN,
 	MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv),
 SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN,
@@ -616,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN,
 SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG,
 	MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv),
 SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG,
-	MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv),
+	MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv),
 SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG,
 	MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv),
 SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG,
@@ -653,29 +654,29 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),
 SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0),
 SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0),
 SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 /* Limiter */
 SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN,
 	MAX98373_LIMITER_EN_SHIFT, 1, 0),

sound/soc/codecs/nau8822.h

@@ -0,0 +1,204 @@
+/*
+ * nau8822.h  --  NAU8822 Soc Audio Codec driver
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.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 __NAU8822_H__
+#define __NAU8822_H__
+
+#define NAU8822_REG_RESET			0x00
+#define NAU8822_REG_POWER_MANAGEMENT_1		0x01
+#define NAU8822_REG_POWER_MANAGEMENT_2		0x02
+#define NAU8822_REG_POWER_MANAGEMENT_3		0x03
+#define NAU8822_REG_AUDIO_INTERFACE		0x04
+#define NAU8822_REG_COMPANDING_CONTROL		0x05
+#define NAU8822_REG_CLOCKING			0x06
+#define NAU8822_REG_ADDITIONAL_CONTROL		0x07
+#define NAU8822_REG_GPIO_CONTROL		0x08
+#define NAU8822_REG_JACK_DETECT_CONTROL_1	0x09
+#define NAU8822_REG_DAC_CONTROL			0x0A
+#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME	0x0B
+#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME	0x0C
+#define NAU8822_REG_JACK_DETECT_CONTROL_2	0x0D
+#define NAU8822_REG_ADC_CONTROL			0x0E
+#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME	0x0F
+#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME	0x10
+#define NAU8822_REG_EQ1				0x12
+#define NAU8822_REG_EQ2				0x13
+#define NAU8822_REG_EQ3				0x14
+#define NAU8822_REG_EQ4				0x15
+#define NAU8822_REG_EQ5				0x16
+#define NAU8822_REG_DAC_LIMITER_1		0x18
+#define NAU8822_REG_DAC_LIMITER_2		0x19
+#define NAU8822_REG_NOTCH_FILTER_1		0x1B
+#define NAU8822_REG_NOTCH_FILTER_2		0x1C
+#define NAU8822_REG_NOTCH_FILTER_3		0x1D
+#define NAU8822_REG_NOTCH_FILTER_4		0x1E
+#define NAU8822_REG_ALC_CONTROL_1		0x20
+#define NAU8822_REG_ALC_CONTROL_2		0x21
+#define NAU8822_REG_ALC_CONTROL_3		0x22
+#define NAU8822_REG_NOISE_GATE			0x23
+#define NAU8822_REG_PLL_N			0x24
+#define NAU8822_REG_PLL_K1			0x25
+#define NAU8822_REG_PLL_K2			0x26
+#define NAU8822_REG_PLL_K3			0x27
+#define NAU8822_REG_3D_CONTROL			0x29
+#define NAU8822_REG_RIGHT_SPEAKER_CONTROL	0x2B
+#define NAU8822_REG_INPUT_CONTROL		0x2C
+#define NAU8822_REG_LEFT_INP_PGA_CONTROL	0x2D
+#define NAU8822_REG_RIGHT_INP_PGA_CONTROL	0x2E
+#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL	0x2F
+#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL	0x30
+#define NAU8822_REG_OUTPUT_CONTROL		0x31
+#define NAU8822_REG_LEFT_MIXER_CONTROL		0x32
+#define NAU8822_REG_RIGHT_MIXER_CONTROL		0x33
+#define NAU8822_REG_LHP_VOLUME			0x34
+#define NAU8822_REG_RHP_VOLUME			0x35
+#define NAU8822_REG_LSPKOUT_VOLUME		0x36
+#define NAU8822_REG_RSPKOUT_VOLUME		0x37
+#define NAU8822_REG_AUX2_MIXER			0x38
+#define NAU8822_REG_AUX1_MIXER			0x39
+#define NAU8822_REG_POWER_MANAGEMENT_4		0x3A
+#define NAU8822_REG_LEFT_TIME_SLOT		0x3B
+#define NAU8822_REG_MISC			0x3C
+#define NAU8822_REG_RIGHT_TIME_SLOT		0x3D
+#define NAU8822_REG_DEVICE_REVISION		0x3E
+#define NAU8822_REG_DEVICE_ID			0x3F
+#define NAU8822_REG_DAC_DITHER			0x41
+#define NAU8822_REG_ALC_ENHANCE_1		0x46
+#define NAU8822_REG_ALC_ENHANCE_2		0x47
+#define NAU8822_REG_192KHZ_SAMPLING		0x48
+#define NAU8822_REG_MISC_CONTROL		0x49
+#define NAU8822_REG_INPUT_TIEOFF		0x4A
+#define NAU8822_REG_POWER_REDUCTION		0x4B
+#define NAU8822_REG_AGC_PEAK2PEAK		0x4C
+#define NAU8822_REG_AGC_PEAK_DETECT		0x4D
+#define NAU8822_REG_AUTOMUTE_CONTROL		0x4E
+#define NAU8822_REG_OUTPUT_TIEOFF		0x4F
+#define NAU8822_REG_MAX_REGISTER		NAU8822_REG_OUTPUT_TIEOFF
+
+/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */
+#define NAU8822_REFIMP_MASK			0x3
+#define NAU8822_REFIMP_80K			0x1
+#define NAU8822_REFIMP_300K			0x2
+#define NAU8822_REFIMP_3K			0x3
+#define NAU8822_IOBUF_EN			(0x1 << 2)
+#define NAU8822_ABIAS_EN			(0x1 << 3)
+
+/* NAU8822_REG_AUDIO_INTERFACE (0x4) */
+#define NAU8822_AIFMT_MASK			(0x3 << 3)
+#define NAU8822_WLEN_MASK			(0x3 << 5)
+#define NAU8822_WLEN_20				(0x1 << 5)
+#define NAU8822_WLEN_24				(0x2 << 5)
+#define NAU8822_WLEN_32				(0x3 << 5)
+#define NAU8822_LRP_MASK			(0x1 << 7)
+#define NAU8822_BCLKP_MASK			(0x1 << 8)
+
+/* NAU8822_REG_COMPANDING_CONTROL (0x5) */
+#define NAU8822_ADDAP_SFT			0
+#define NAU8822_ADCCM_SFT			1
+#define NAU8822_DACCM_SFT			3
+
+/* NAU8822_REG_CLOCKING (0x6) */
+#define NAU8822_CLKIOEN_MASK			0x1
+#define NAU8822_MCLKSEL_SFT			5
+#define NAU8822_MCLKSEL_MASK			(0x7 << 5)
+#define NAU8822_BCLKSEL_SFT			2
+#define NAU8822_BCLKSEL_MASK			(0x7 << 2)
+#define NAU8822_CLKM_MASK			(0x1 << 8)
+#define NAU8822_CLKM_MCLK			(0x0 << 8)
+#define NAU8822_CLKM_PLL			(0x1 << 8)
+
+/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */
+#define NAU8822_SMPLR_SFT			1
+#define NAU8822_SMPLR_MASK			(0x7 << 1)
+#define NAU8822_SMPLR_48K			(0x0 << 1)
+#define NAU8822_SMPLR_32K			(0x1 << 1)
+#define NAU8822_SMPLR_24K			(0x2 << 1)
+#define NAU8822_SMPLR_16K			(0x3 << 1)
+#define NAU8822_SMPLR_12K			(0x4 << 1)
+#define NAU8822_SMPLR_8K			(0x5 << 1)
+
+/* NAU8822_REG_EQ1 (0x12) */
+#define NAU8822_EQ1GC_SFT			0
+#define NAU8822_EQ1CF_SFT			5
+#define NAU8822_EQM_SFT				8
+
+/* NAU8822_REG_EQ2 (0x13) */
+#define NAU8822_EQ2GC_SFT			0
+#define NAU8822_EQ2CF_SFT			5
+#define NAU8822_EQ2BW_SFT			8
+
+/* NAU8822_REG_EQ3 (0x14) */
+#define NAU8822_EQ3GC_SFT			0
+#define NAU8822_EQ3CF_SFT			5
+#define NAU8822_EQ3BW_SFT			8
+
+/* NAU8822_REG_EQ4 (0x15) */
+#define NAU8822_EQ4GC_SFT			0
+#define NAU8822_EQ4CF_SFT			5
+#define NAU8822_EQ4BW_SFT			8
+
+/* NAU8822_REG_EQ5 (0x16) */
+#define NAU8822_EQ5GC_SFT			0
+#define NAU8822_EQ5CF_SFT			5
+
+/* NAU8822_REG_ALC_CONTROL_1 (0x20) */
+#define NAU8822_ALCMINGAIN_SFT			0
+#define NAU8822_ALCMXGAIN_SFT			3
+#define NAU8822_ALCEN_SFT			7
+
+/* NAU8822_REG_ALC_CONTROL_2 (0x21) */
+#define NAU8822_ALCSL_SFT			0
+#define NAU8822_ALCHT_SFT			4
+
+/* NAU8822_REG_ALC_CONTROL_3 (0x22) */
+#define NAU8822_ALCATK_SFT			0
+#define NAU8822_ALCDCY_SFT			4
+#define NAU8822_ALCM_SFT			8
+
+/* NAU8822_REG_PLL_N (0x24) */
+#define NAU8822_PLLMCLK_DIV2			(0x1 << 4)
+#define NAU8822_PLLN_MASK			0xF
+
+#define NAU8822_PLLK1_SFT			18
+#define NAU8822_PLLK1_MASK			0x3F
+
+/* NAU8822_REG_PLL_K2 (0x26) */
+#define NAU8822_PLLK2_SFT			9
+#define NAU8822_PLLK2_MASK			0x1FF
+
+/* NAU8822_REG_PLL_K3 (0x27) */
+#define NAU8822_PLLK3_MASK			0x1FF
+
+/* System Clock Source */
+enum {
+	NAU8822_CLK_MCLK,
+	NAU8822_CLK_PLL,
+};
+
+struct nau8822_pll {
+	int pre_factor;
+	int mclk_scaler;
+	int pll_frac;
+	int pll_int;
+};
+
+/* Codec Private Data */
+struct nau8822 {
+	struct device *dev;
+	struct regmap *regmap;
+	int mclk_idx;
+	struct nau8822_pll pll;
+	int sysclk;
+	int div_id;
+};
+
+#endif	/* __NAU8822_H__ */

sound/soc/codecs/pcm186x.c

@@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		priv->tdm_offset += 1;
-		/* Fall through... DSP_A uses the same basic config as DSP_B
+		/* fall through */
+		/* DSP_A uses the same basic config as DSP_B
 		 * except we need to shift the TDM output by one BCK cycle
 		 */
 	case SND_SOC_DAIFMT_DSP_B:

sound/soc/codecs/pcm3060-i2c.c

@@ -0,0 +1,60 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 I2C driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct pcm3060_priv *priv;
+
+	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, priv);
+
+	priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap);
+	if (IS_ERR(priv->regmap))
+		return PTR_ERR(priv->regmap);
+
+	return pcm3060_probe(&i2c->dev);
+}
+
+static const struct i2c_device_id pcm3060_i2c_id[] = {
+	{ .name = "pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+	{ .compatible = "ti,pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct i2c_driver pcm3060_i2c_driver = {
+	.driver = {
+		.name = "pcm3060",
+#ifdef CONFIG_OF
+		.of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+	},
+	.id_table = pcm3060_i2c_id,
+	.probe = pcm3060_i2c_probe,
+};
+
+module_i2c_driver(pcm3060_i2c_driver);
+
+MODULE_DESCRIPTION("PCM3060 I2C driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060-spi.c

@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 SPI driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_spi_probe(struct spi_device *spi)
+{
+	struct pcm3060_priv *priv;
+
+	priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, priv);
+
+	priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap);
+	if (IS_ERR(priv->regmap))
+		return PTR_ERR(priv->regmap);
+
+	return pcm3060_probe(&spi->dev);
+}
+
+static const struct spi_device_id pcm3060_spi_id[] = {
+	{ .name = "pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, pcm3060_spi_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+	{ .compatible = "ti,pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct spi_driver pcm3060_spi_driver = {
+	.driver = {
+		.name = "pcm3060",
+#ifdef CONFIG_OF
+		.of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+	},
+	.id_table = pcm3060_spi_id,
+	.probe = pcm3060_spi_probe,
+};
+
+module_spi_driver(pcm3060_spi_driver);
+
+MODULE_DESCRIPTION("PCM3060 SPI driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060.c

@@ -0,0 +1,295 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 codec driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "pcm3060.h"
+
+/* dai */
+
+static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			      unsigned int freq, int dir)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+
+	if (dir != SND_SOC_CLOCK_IN) {
+		dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir);
+		return -EINVAL;
+	}
+
+	priv->dai[dai->id].sclk_freq = freq;
+
+	return 0;
+}
+
+static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+	unsigned int reg;
+	unsigned int val;
+
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		priv->dai[dai->id].is_master = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		priv->dai[dai->id].is_master = false;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val = PCM3060_REG_FMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val = PCM3060_REG_FMT_RJ;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val = PCM3060_REG_FMT_LJ;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	if (dai->id == PCM3060_DAI_ID_DAC)
+		reg = PCM3060_REG67;
+	else
+		reg = PCM3060_REG72;
+
+	regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val);
+
+	return 0;
+}
+
+static int pcm3060_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+	unsigned int rate;
+	unsigned int ratio;
+	unsigned int reg;
+	unsigned int val;
+
+	if (!priv->dai[dai->id].is_master) {
+		val = PCM3060_REG_MS_S;
+		goto val_ready;
+	}
+
+	rate = params_rate(params);
+	if (!rate) {
+		dev_err(comp->dev, "rate is not configured\n");
+		return -EINVAL;
+	}
+
+	ratio = priv->dai[dai->id].sclk_freq / rate;
+
+	switch (ratio) {
+	case 768:
+		val = PCM3060_REG_MS_M768;
+		break;
+	case 512:
+		val = PCM3060_REG_MS_M512;
+		break;
+	case 384:
+		val = PCM3060_REG_MS_M384;
+		break;
+	case 256:
+		val = PCM3060_REG_MS_M256;
+		break;
+	case 192:
+		val = PCM3060_REG_MS_M192;
+		break;
+	case 128:
+		val = PCM3060_REG_MS_M128;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported ratio: %d\n", ratio);
+		return -EINVAL;
+	}
+
+val_ready:
+	if (dai->id == PCM3060_DAI_ID_DAC)
+		reg = PCM3060_REG67;
+	else
+		reg = PCM3060_REG72;
+
+	regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops pcm3060_dai_ops = {
+	.set_sysclk = pcm3060_set_sysclk,
+	.set_fmt = pcm3060_set_fmt,
+	.hw_params = pcm3060_hw_params,
+};
+
+#define PCM3060_DAI_RATES_ADC	(SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \
+				 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+				 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PCM3060_DAI_RATES_DAC	(PCM3060_DAI_RATES_ADC | \
+				 SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+static struct snd_soc_dai_driver pcm3060_dai[] = {
+	{
+		.name = "pcm3060-dac",
+		.id = PCM3060_DAI_ID_DAC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = PCM3060_DAI_RATES_DAC,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops = &pcm3060_dai_ops,
+	},
+	{
+		.name = "pcm3060-adc",
+		.id = PCM3060_DAI_ID_ADC,
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = PCM3060_DAI_RATES_ADC,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops = &pcm3060_dai_ops,
+	},
+};
+
+/* dapm */
+
+static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1);
+
+static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
+	SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume",
+			       PCM3060_REG65, PCM3060_REG66, 0,
+			       PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX,
+			       0, pcm3060_dapm_tlv),
+	SOC_DOUBLE("Master Playback Switch", PCM3060_REG68,
+		   PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1),
+
+	SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume",
+			       PCM3060_REG70, PCM3060_REG71, 0,
+			       PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX,
+			       0, pcm3060_dapm_tlv),
+	SOC_DOUBLE("Master Capture Switch", PCM3060_REG73,
+		   PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1),
+};
+
+static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("OUTL+"),
+	SND_SOC_DAPM_OUTPUT("OUTR+"),
+	SND_SOC_DAPM_OUTPUT("OUTL-"),
+	SND_SOC_DAPM_OUTPUT("OUTR-"),
+
+	SND_SOC_DAPM_INPUT("INL"),
+	SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
+	{ "OUTL+", NULL, "Playback" },
+	{ "OUTR+", NULL, "Playback" },
+	{ "OUTL-", NULL, "Playback" },
+	{ "OUTR-", NULL, "Playback" },
+
+	{ "Capture", NULL, "INL" },
+	{ "Capture", NULL, "INR" },
+};
+
+/* soc component */
+
+static const struct snd_soc_component_driver pcm3060_soc_comp_driver = {
+	.controls = pcm3060_dapm_controls,
+	.num_controls = ARRAY_SIZE(pcm3060_dapm_controls),
+	.dapm_widgets = pcm3060_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets),
+	.dapm_routes = pcm3060_dapm_map,
+	.num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map),
+};
+
+/* regmap */
+
+static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg)
+{
+	return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_readable(struct device *dev, unsigned int reg)
+{
+	return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg)
+{
+	/* PCM3060_REG64 is volatile */
+	return (reg == PCM3060_REG64);
+}
+
+static const struct reg_default pcm3060_reg_defaults[] = {
+	{ PCM3060_REG64,  0xF0 },
+	{ PCM3060_REG65,  0xFF },
+	{ PCM3060_REG66,  0xFF },
+	{ PCM3060_REG67,  0x00 },
+	{ PCM3060_REG68,  0x00 },
+	{ PCM3060_REG69,  0x00 },
+	{ PCM3060_REG70,  0xD7 },
+	{ PCM3060_REG71,  0xD7 },
+	{ PCM3060_REG72,  0x00 },
+	{ PCM3060_REG73,  0x00 },
+};
+
+const struct regmap_config pcm3060_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.writeable_reg = pcm3060_reg_writeable,
+	.readable_reg = pcm3060_reg_readable,
+	.volatile_reg = pcm3060_reg_volatile,
+	.max_register = PCM3060_REG73,
+	.reg_defaults = pcm3060_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(pcm3060_regmap);
+
+/* device */
+
+int pcm3060_probe(struct device *dev)
+{
+	int rc;
+
+	rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver,
+					     pcm3060_dai,
+					     ARRAY_SIZE(pcm3060_dai));
+	if (rc) {
+		dev_err(dev, "failed to register component, rc=%d\n", rc);
+		return rc;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL(pcm3060_probe);
+
+MODULE_DESCRIPTION("PCM3060 codec driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060.h

@@ -0,0 +1,88 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * PCM3060 codec driver
+ *
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ */
+
+#ifndef _SND_SOC_PCM3060_H
+#define _SND_SOC_PCM3060_H
+
+#include <linux/device.h>
+#include <linux/regmap.h>
+
+extern const struct regmap_config pcm3060_regmap;
+
+#define PCM3060_DAI_ID_DAC	0
+#define PCM3060_DAI_ID_ADC	1
+#define PCM3060_DAI_IDS_NUM	2
+
+struct pcm3060_priv_dai {
+	bool is_master;
+	unsigned int sclk_freq;
+};
+
+struct pcm3060_priv {
+	struct regmap *regmap;
+	struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM];
+};
+
+int pcm3060_probe(struct device *dev);
+int pcm3060_remove(struct device *dev);
+
+/* registers */
+
+#define PCM3060_REG64			0x40
+#define PCM3060_REG_MRST		0x80
+#define PCM3060_REG_SRST		0x40
+#define PCM3060_REG_ADPSV		0x20
+#define PCM3060_REG_DAPSV		0x10
+#define PCM3060_REG_SE			0x01
+
+#define PCM3060_REG65			0x41
+#define PCM3060_REG66			0x42
+#define PCM3060_REG_AT2_MIN		0x36
+#define PCM3060_REG_AT2_MAX		0xFF
+
+#define PCM3060_REG67			0x43
+#define PCM3060_REG72			0x48
+#define PCM3060_REG_CSEL		0x80
+#define PCM3060_REG_MASK_MS		0x70
+#define PCM3060_REG_MS_S		0x00
+#define PCM3060_REG_MS_M768		(0x01 << 4)
+#define PCM3060_REG_MS_M512		(0x02 << 4)
+#define PCM3060_REG_MS_M384		(0x03 << 4)
+#define PCM3060_REG_MS_M256		(0x04 << 4)
+#define PCM3060_REG_MS_M192		(0x05 << 4)
+#define PCM3060_REG_MS_M128		(0x06 << 4)
+#define PCM3060_REG_MASK_FMT		0x03
+#define PCM3060_REG_FMT_I2S		0x00
+#define PCM3060_REG_FMT_LJ		0x01
+#define PCM3060_REG_FMT_RJ		0x02
+
+#define PCM3060_REG68			0x44
+#define PCM3060_REG_OVER		0x40
+#define PCM3060_REG_DREV2		0x04
+#define PCM3060_REG_SHIFT_MUT21	0x00
+#define PCM3060_REG_SHIFT_MUT22	0x01
+
+#define PCM3060_REG69			0x45
+#define PCM3060_REG_FLT		0x80
+#define PCM3060_REG_MASK_DMF		0x60
+#define PCM3060_REG_DMC		0x10
+#define PCM3060_REG_ZREV		0x02
+#define PCM3060_REG_AZRO		0x01
+
+#define PCM3060_REG70			0x46
+#define PCM3060_REG71			0x47
+#define PCM3060_REG_AT1_MIN		0x0E
+#define PCM3060_REG_AT1_MAX		0xFF
+
+#define PCM3060_REG73			0x49
+#define PCM3060_REG_ZCDD		0x10
+#define PCM3060_REG_BYP		0x08
+#define PCM3060_REG_DREV1		0x04
+#define PCM3060_REG_SHIFT_MUT11	0x00
+#define PCM3060_REG_SHIFT_MUT12	0x01
+
+#endif /* _SND_SOC_PCM3060_H */

sound/soc/codecs/pcm3168a.c

@@ -33,6 +33,8 @@
 #define PCM3168A_FMT_RIGHT_J_16		0x3
 #define PCM3168A_FMT_DSP_A		0x4
 #define PCM3168A_FMT_DSP_B		0x5
+#define PCM3168A_FMT_I2S_TDM		0x6
+#define PCM3168A_FMT_LEFT_J_TDM		0x7
 #define PCM3168A_FMT_DSP_MASK		0x4
 
 #define PCM3168A_NUM_SUPPLIES 6
@@ -401,9 +403,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	bool tx, master_mode;
 	u32 val, mask, shift, reg;
 	unsigned int rate, fmt, ratio, max_ratio;
+	unsigned int chan;
 	int i, min_frame_size;
 
 	rate = params_rate(params);
+	chan = params_channels(params);
 
 	ratio = pcm3168a->sysclk / rate;
 
@@ -456,6 +460,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	/* for TDM */
+	if (chan > 2) {
+		switch (fmt) {
+		case PCM3168A_FMT_I2S:
+			fmt = PCM3168A_FMT_I2S_TDM;
+			break;
+		case PCM3168A_FMT_LEFT_J:
+			fmt = PCM3168A_FMT_LEFT_J_TDM;
+			break;
+		default:
+			dev_err(component->dev, "TDM is supported under I2S/Left_J only\n");
+			return -EINVAL;
+		}
+	}
+
 	if (master_mode)
 		val = ((i + 1) << shift);
 	else
@@ -476,7 +495,69 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int pcm3168a_startup(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int fmt;
+	unsigned int sample_min;
+	unsigned int channel_max;
+
+	if (tx)
+		fmt = pcm3168a->dac_fmt;
+	else
+		fmt = pcm3168a->adc_fmt;
+
+	/*
+	 * Available Data Bits
+	 *
+	 * RIGHT_J : 24 / 16
+	 * LEFT_J  : 24
+	 * I2S     : 24
+	 *
+	 * TDM available
+	 *
+	 * I2S
+	 * LEFT_J
+	 */
+	switch (fmt) {
+	case PCM3168A_FMT_RIGHT_J:
+		sample_min  = 16;
+		channel_max =  2;
+		break;
+	case PCM3168A_FMT_LEFT_J:
+		sample_min  = 24;
+		if (tx)
+			channel_max = 8;
+		else
+			channel_max = 6;
+		break;
+	case PCM3168A_FMT_I2S:
+		sample_min  = 24;
+		if (tx)
+			channel_max = 8;
+		else
+			channel_max = 6;
+		break;
+	default:
+		sample_min  = 24;
+		channel_max =  2;
+	}
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+				     sample_min, 32);
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS,
+				     2, channel_max);
+
+	return 0;
+}
 static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
+	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_dac,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params,
@@ -484,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
 };
 
 static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
+	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_adc,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params

sound/soc/codecs/rt274.c

@@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 		break;
 	default:
 		dev_warn(component->dev, "invalid pll source, use BCLK\n");
+		/* fall through */
 	case RT274_PLL2_S_BCLK:
 		snd_soc_component_update_bits(component, RT274_PLL2_CTRL,
 				RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK);
@@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 			break;
 		default:
 			dev_warn(component->dev, "invalid freq_in, assume 4.8M\n");
+			/* fall through */
 		case 100:
 			snd_soc_component_write(component, 0x7a, 0xaab6);
 			snd_soc_component_write(component, 0x7b, 0x0301);

sound/soc/codecs/rt5651.c

@@ -10,7 +10,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>

sound/soc/codecs/rt5663.c

@@ -72,6 +72,7 @@ struct rt5663_priv {
 static const struct reg_sequence rt5663_patch_list[] = {
 	{ 0x002a, 0x8020 },
 	{ 0x0086, 0x0028 },
+	{ 0x0100, 0xa020 },
 	{ 0x0117, 0x0f28 },
 	{ 0x02fb, 0x8089 },
 };
@@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x00fd, 0x0001 },
 	{ 0x00fe, 0x10ec },
 	{ 0x00ff, 0x6406 },
-	{ 0x0100, 0xa0a0 },
+	{ 0x0100, 0xa020 },
 	{ 0x0108, 0x4444 },
 	{ 0x0109, 0x4444 },
 	{ 0x010a, 0xaaaa },
@@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				0x8000);
 			snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000,
 				0x3000);
+			snd_soc_component_update_bits(component,
+				RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080);
 		}
 		break;
 
@@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN);
 			snd_soc_component_update_bits(component,
 				RT5663_DACREF_LDO, 0x3e0e, 0);
+			snd_soc_component_update_bits(component,
+				RT5663_DIG_VOL_ZCD, 0x00c0, 0);
 		}
 		break;
 

sound/soc/codecs/rt5668.c

@@ -2587,17 +2587,10 @@ static int rt5668_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
+	return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
 			rt5668_dai, ARRAY_SIZE(rt5668_dai));
 }
 
-static int rt5668_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-
-	return 0;
-}
-
 static void rt5668_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt5668_priv *rt5668 = i2c_get_clientdata(client);
@@ -2628,7 +2621,6 @@ static struct i2c_driver rt5668_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5668_acpi_match),
 	},
 	.probe = rt5668_i2c_probe,
-	.remove = rt5668_i2c_remove,
 	.shutdown = rt5668_i2c_shutdown,
 	.id_table = rt5668_i2c_id,
 };

sound/soc/codecs/rt5670.c

@@ -2875,6 +2875,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
 						 RT5670_DEV_GPIO |
 						 RT5670_JD_MODE1),
 	},
+	{
+		.callback = rt5670_quirk_cb,
+		.ident = "Lenovo Thinkpad Tablet 8",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+		},
+		.driver_data = (unsigned long *)(RT5670_DMIC_EN |
+						 RT5670_DMIC2_INR |
+						 RT5670_DEV_GPIO |
+						 RT5670_JD_MODE1),
+	},
 	{
 		.callback = rt5670_quirk_cb,
 		.ident = "Lenovo Thinkpad Tablet 10",

sound/soc/codecs/rt5677-spi.c

@@ -18,7 +18,6 @@
 #include <linux/interrupt.h>
 #include <linux/irq.h>
 #include <linux/slab.h>
-#include <linux/gpio.h>
 #include <linux/sched.h>
 #include <linux/uaccess.h>
 #include <linux/regulator/consumer.h>

sound/soc/codecs/rt5682.c

@@ -67,7 +67,8 @@ struct rt5682_priv {
 };
 
 static const struct reg_sequence patch_list[] = {
-	{0x01c1, 0x1000},
+	{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
+	{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
 };
 
 static const struct reg_default rt5682_reg[] = {
@@ -1432,6 +1433,28 @@ static const struct snd_kcontrol_new hpor_switch =
 	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
 					RT5682_R_MUTE_SFT, 1, 1);
 
+static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -1444,10 +1467,10 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_HP_LOGIC_CTRL_2, 0x0012);
 		snd_soc_component_write(component,
 			RT5682_HP_CTRL_2, 0x6000);
-		snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1,
-			RT5682_NG2_EN_MASK, RT5682_NG2_EN);
 		snd_soc_component_update_bits(component,
 			RT5682_DEPOP_1, 0x60, 0x60);
+		snd_soc_component_update_bits(component,
+			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080);
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
@@ -1455,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_DEPOP_1, 0x60, 0x0);
 		snd_soc_component_write(component,
 			RT5682_HP_CTRL_2, 0x0000);
+		snd_soc_component_update_bits(component,
+			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000);
 		break;
 
 	default:
@@ -1718,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1,
 		RT5682_PWR_HA_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1,
-		RT5682_PUMP_EN_SFT, 0, NULL, 0),
+		RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event,
+		SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
 		RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
 
@@ -1879,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
 	{"HP Amp", NULL, "Charge Pump"},
 	{"HP Amp", NULL, "CLKDET SYS"},
 	{"HP Amp", NULL, "CBJ Power"},
+	{"HP Amp", NULL, "Vref1"},
 	{"HP Amp", NULL, "Vref2"},
 	{"HPOL Playback", "Switch", "HP Amp"},
 	{"HPOR Playback", "Switch", "HP Amp"},
@@ -2446,30 +2473,23 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 	mutex_lock(&rt5682->calibrate_mutex);
 
 	rt5682_reset(rt5682->regmap);
-	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
 	usleep_range(15000, 20000);
-	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf);
-	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
-	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001);
-	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
-	regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080);
-	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040);
-	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300);
+	regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100);
+	regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800);
 	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000);
-	regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000);
-	regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26);
-	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005);
 	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c);
 	regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d);
-	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f);
-	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321);
 	regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1);
 	regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311);
-	regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000);
-	regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
 
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
 
@@ -2485,8 +2505,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 		pr_err("HP Calibration Failure\n");
 
 	/* restore settings */
-	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080);
+	regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000);
 	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
 
 	mutex_unlock(&rt5682->calibrate_mutex);
 
@@ -2560,7 +2584,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	rt5682_calibrate(rt5682);
 
-	ret = regmap_register_patch(rt5682->regmap, patch_list,
+	ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
 				    ARRAY_SIZE(patch_list));
 	if (ret != 0)
 		dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
@@ -2614,6 +2638,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 			RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK,
 			RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1);
 	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+	regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+			RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+	regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+			RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
 
 	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
 				rt5682_jack_detect_handler);
@@ -2631,11 +2659,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return devm_snd_soc_register_component(&i2c->dev,
-			&soc_component_dev_rt5682,
+	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682,
 			rt5682_dai, ARRAY_SIZE(rt5682_dai));
 }
 
+static int rt5682_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_component(&i2c->dev);
+
+	return 0;
+}
+
 static void rt5682_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
@@ -2666,6 +2700,7 @@ static struct i2c_driver rt5682_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
 	},
 	.probe = rt5682_i2c_probe,
+	.remove = rt5682_i2c_remove,
 	.shutdown = rt5682_i2c_shutdown,
 	.id_table = rt5682_i2c_id,
 };

sound/soc/codecs/rt5682.h

@@ -1214,6 +1214,20 @@
 #define RT5682_JDH_NO_PLUG			(0x1 << 4)
 #define RT5682_JDH_PLUG				(0x0 << 4)
 
+/* Bias current control 8 (0x0111) */
+#define RT5682_HPA_CP_BIAS_CTRL_MASK			(0x3 << 2)
+#define RT5682_HPA_CP_BIAS_2UA			(0x0 << 2)
+#define RT5682_HPA_CP_BIAS_3UA			(0x1 << 2)
+#define RT5682_HPA_CP_BIAS_4UA			(0x2 << 2)
+#define RT5682_HPA_CP_BIAS_6UA			(0x3 << 2)
+
+/* Charge Pump Internal Register1 (0x0125) */
+#define RT5682_CP_CLK_HP_MASK			(0x3 << 4)
+#define RT5682_CP_CLK_HP_100KHZ			(0x0 << 4)
+#define RT5682_CP_CLK_HP_200KHZ			(0x1 << 4)
+#define RT5682_CP_CLK_HP_300KHZ			(0x2 << 4)
+#define RT5682_CP_CLK_HP_600KHZ			(0x3 << 4)
+
 /* Chopper and Clock control for DAC (0x013a)*/
 #define RT5682_CKXEN_DAC1_MASK			(0x1 << 13)
 #define RT5682_CKXEN_DAC1_SFT			13

sound/soc/codecs/sgtl5000.c

@@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
 	 * Searching for a suitable index solving this formula:
 	 * idx = 40 * log10(vag_val / lo_cagcntrl) + 15
 	 */
-	vol_quot = (vag * 100) / lo_vag;
+	vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
 	lo_vol = 0;
 	for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
 		if (vol_quot >= vol_quot_table[i])

sound/soc/codecs/sta32x.c

@@ -21,6 +21,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
@@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = {
 /* codec private data */
 struct sta32x_priv {
 	struct regmap *regmap;
+	struct clk *xti_clk;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
 	struct snd_soc_component *component;
 	struct sta32x_platform_data *pdata;
@@ -882,6 +884,15 @@ static int sta32x_probe(struct snd_soc_component *component)
 
 	sta32x->component = component;
 
+	if (sta32x->xti_clk) {
+		ret = clk_prepare_enable(sta32x->xti_clk);
+		if (ret != 0) {
+			dev_err(component->dev,
+				"Failed to enable clock: %d\n", ret);
+			return ret;
+		}
+	}
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
 				    sta32x->supplies);
 	if (ret != 0) {
@@ -984,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component)
 
 	sta32x_watchdog_stop(sta32x);
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+	if (sta32x->xti_clk)
+		clk_disable_unprepare(sta32x->xti_clk);
 }
 
 static const struct snd_soc_component_driver sta32x_component = {
@@ -1041,6 +1055,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x)
 	of_property_read_u8(np, "st,ch3-output-mapping",
 			    &pdata->ch3_output_mapping);
 
+	if (of_get_property(np, "st,fault-detect-recovery", NULL))
+		pdata->fault_detect_recovery = 1;
 	if (of_get_property(np, "st,thermal-warning-recovery", NULL))
 		pdata->thermal_warning_recovery = 1;
 	if (of_get_property(np, "st,thermal-warning-adjustment", NULL))
@@ -1098,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c,
 	}
 #endif
 
+	/* Clock */
+	sta32x->xti_clk = devm_clk_get(dev, "xti");
+	if (IS_ERR(sta32x->xti_clk)) {
+		ret = PTR_ERR(sta32x->xti_clk);
+
+		if (ret == -EPROBE_DEFER)
+			return ret;
+
+		sta32x->xti_clk = NULL;
+	}
+
 	/* GPIOs */
 	sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
 						       GPIOD_OUT_LOW);

sound/soc/codecs/tas5720.c

@@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 				    int slots, int slot_width)
 {
 	struct snd_soc_component *component = dai->component;
+	struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component);
 	unsigned int first_slot;
 	int ret;
 
@@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	if (ret < 0)
 		goto error_snd_soc_component_update_bits;
 
+	/* Configure TDM slot width. This is only applicable to TAS5722. */
+	switch (tas5720->devtype) {
+	case TAS5722:
+		ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+						    TAS5722_TDM_SLOT_16B,
+						    slot_width == 16 ?
+						    TAS5722_TDM_SLOT_16B : 0);
+		if (ret < 0)
+			goto error_snd_soc_component_update_bits;
+		break;
+	default:
+		break;
+	}
+
 	return 0;
 
 error_snd_soc_component_update_bits:
@@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
 );
 
 /*
- * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
- * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
- * as per device datasheet.
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps
+ * depending on the device. Note that setting the gain below -100 dB
+ * (register value <0x7) is effectively a MUTE as per device datasheet.
+ *
+ * Note that for the TAS5722 the digital volume controls are actually split
+ * over two registers, so we need custom getters/setters for access.
  */
-static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0);
+
+static int tas5722_volume_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	unsigned int val;
+
+	snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val);
+	ucontrol->value.integer.value[0] = val << 1;
+
+	snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val);
+	ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB;
+
+	return 0;
+}
+
+static int tas5722_volume_set(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	unsigned int sel = ucontrol->value.integer.value[0];
+
+	snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1);
+	snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+				      TAS5722_VOL_CONTROL_LSB, sel);
+
+	return 0;
+}
 
 static const struct snd_kcontrol_new tas5720_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
-		       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+		       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv),
+	SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+		       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_kcontrol_new tas5722_snd_controls[] = {
+	SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume",
+			   0, 0, 511, 0,
+			   tas5722_volume_get, tas5722_volume_set,
+			   tas5722_dac_tlv),
 	SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
 		       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
 };
@@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static const struct snd_soc_component_driver soc_component_dev_tas5722 = {
+	.probe = tas5720_codec_probe,
+	.remove = tas5720_codec_remove,
+	.suspend = tas5720_suspend,
+	.resume = tas5720_resume,
+	.controls = tas5722_snd_controls,
+	.num_controls = ARRAY_SIZE(tas5722_snd_controls),
+	.dapm_widgets = tas5720_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+	.dapm_routes = tas5720_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
 /* PCM rates supported by the TAS5720 driver */
 #define TAS5720_RATES	(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
 			 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
@@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client,
 
 	dev_set_drvdata(dev, data);
 
-	ret = devm_snd_soc_register_component(&client->dev,
-				     &soc_component_dev_tas5720,
-				     tas5720_dai, ARRAY_SIZE(tas5720_dai));
+	switch (id->driver_data) {
+	case TAS5720:
+		ret = devm_snd_soc_register_component(&client->dev,
+					&soc_component_dev_tas5720,
+					tas5720_dai,
+					ARRAY_SIZE(tas5720_dai));
+		break;
+	case TAS5722:
+		ret = devm_snd_soc_register_component(&client->dev,
+					&soc_component_dev_tas5722,
+					tas5720_dai,
+					ARRAY_SIZE(tas5720_dai));
+		break;
+	default:
+		dev_err(dev, "unexpected private driver data\n");
+		return -EINVAL;
+	}
 	if (ret < 0) {
 		dev_err(dev, "failed to register component: %d\n", ret);
 		return ret;

sound/soc/codecs/tas6424.c

@@ -41,6 +41,7 @@ struct tas6424_data {
 	struct regmap *regmap;
 	struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES];
 	struct delayed_work fault_check_work;
+	unsigned int last_cfault;
 	unsigned int last_fault1;
 	unsigned int last_fault2;
 	unsigned int last_warn;
@@ -406,9 +407,54 @@ static void tas6424_fault_check_work(struct work_struct *work)
 	unsigned int reg;
 	int ret;
 
+	ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, &reg);
+	if (ret < 0) {
+		dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret);
+		goto out;
+	}
+
+	if (!reg) {
+		tas6424->last_cfault = reg;
+		goto check_global_fault1_reg;
+	}
+
+	/*
+	 * Only flag errors once for a given occurrence. This is needed as
+	 * the TAS6424 will take time clearing the fault condition internally
+	 * during which we don't want to bombard the system with the same
+	 * error message over and over.
+	 */
+	if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1))
+		dev_crit(dev, "experienced a channel 1 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2))
+		dev_crit(dev, "experienced a channel 2 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3))
+		dev_crit(dev, "experienced a channel 3 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4))
+		dev_crit(dev, "experienced a channel 4 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1))
+		dev_crit(dev, "experienced a channel 1 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2))
+		dev_crit(dev, "experienced a channel 2 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3))
+		dev_crit(dev, "experienced a channel 3 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4))
+		dev_crit(dev, "experienced a channel 4 DC fault\n");
+
+	/* Store current fault1 value so we can detect any changes next time */
+	tas6424->last_cfault = reg;
+
+check_global_fault1_reg:
 	ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg);
 	if (ret < 0) {
-		dev_err(dev, "failed to read FAULT1 register: %d\n", ret);
+		dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret);
 		goto out;
 	}
 
@@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work)
 		goto check_global_fault2_reg;
 	}
 
-	/*
-	 * Only flag errors once for a given occurrence. This is needed as
-	 * the TAS6424 will take time clearing the fault condition internally
-	 * during which we don't want to bombard the system with the same
-	 * error message over and over.
-	 */
 	if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV))
 		dev_crit(dev, "experienced a PVDD overvoltage fault\n");
 
@@ -453,7 +493,7 @@ static void tas6424_fault_check_work(struct work_struct *work)
 check_global_fault2_reg:
 	ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, &reg);
 	if (ret < 0) {
-		dev_err(dev, "failed to read FAULT2 register: %d\n", ret);
+		dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret);
 		goto out;
 	}
 
@@ -530,7 +570,7 @@ static void tas6424_fault_check_work(struct work_struct *work)
 	/* Store current warn value so we can detect any changes next time */
 	tas6424->last_warn = reg;
 
-	/* Clear any faults by toggling the CLEAR_FAULT control bit */
+	/* Clear any warnings by toggling the CLEAR_FAULT control bit */
 	ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3,
 				TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT);
 	if (ret < 0)

sound/soc/codecs/tas6424.h

@@ -115,6 +115,16 @@
 #define TAS6424_LDGBYPASS_SHIFT		0
 #define TAS6424_LDGBYPASS_MASK		BIT(TAS6424_LDGBYPASS_SHIFT)
 
+/* TAS6424_GLOB_FAULT1_REG */
+#define TAS6424_FAULT_OC_CH1		BIT(7)
+#define TAS6424_FAULT_OC_CH2		BIT(6)
+#define TAS6424_FAULT_OC_CH3		BIT(5)
+#define TAS6424_FAULT_OC_CH4		BIT(4)
+#define TAS6424_FAULT_DC_CH1		BIT(3)
+#define TAS6424_FAULT_DC_CH2		BIT(2)
+#define TAS6424_FAULT_DC_CH3		BIT(1)
+#define TAS6424_FAULT_DC_CH4		BIT(0)
+
 /* TAS6424_GLOB_FAULT1_REG */
 #define TAS6424_FAULT_CLOCK		BIT(4)
 #define TAS6424_FAULT_PVDD_OV		BIT(3)

sound/soc/codecs/tlv320aic31xx.c

@@ -167,6 +167,7 @@ struct aic31xx_priv {
 	u8 p_div;
 	int rate_div_line;
 	bool master_dapm_route_applied;
+	int irq;
 };
 
 struct aic31xx_rate_divs {
@@ -1391,6 +1392,69 @@ static const struct acpi_device_id aic31xx_acpi_match[] = {
 MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
 #endif
 
+static irqreturn_t aic31xx_irq(int irq, void *data)
+{
+	struct aic31xx_priv *aic31xx = data;
+	struct device *dev = aic31xx->dev;
+	unsigned int value;
+	bool handled = false;
+	int ret;
+
+	ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value);
+	if (ret) {
+		dev_err(dev, "Failed to read interrupt mask: %d\n", ret);
+		goto exit;
+	}
+
+	if (value)
+		handled = true;
+	else
+		goto read_overflow;
+
+	if (value & AIC31XX_HPLSCDETECT)
+		dev_err(dev, "Short circuit on Left output is detected\n");
+	if (value & AIC31XX_HPRSCDETECT)
+		dev_err(dev, "Short circuit on Right output is detected\n");
+	if (value & ~(AIC31XX_HPLSCDETECT |
+		      AIC31XX_HPRSCDETECT))
+		dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value);
+
+read_overflow:
+	ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value);
+	if (ret) {
+		dev_err(dev, "Failed to read overflow flag: %d\n", ret);
+		goto exit;
+	}
+
+	if (value)
+		handled = true;
+	else
+		goto exit;
+
+	if (value & AIC31XX_DAC_OF_LEFT)
+		dev_warn(dev, "Left-channel DAC overflow has occurred\n");
+	if (value & AIC31XX_DAC_OF_RIGHT)
+		dev_warn(dev, "Right-channel DAC overflow has occurred\n");
+	if (value & AIC31XX_DAC_OF_SHIFTER)
+		dev_warn(dev, "DAC barrel shifter overflow has occurred\n");
+	if (value & AIC31XX_ADC_OF)
+		dev_warn(dev, "ADC overflow has occurred\n");
+	if (value & AIC31XX_ADC_OF_SHIFTER)
+		dev_warn(dev, "ADC barrel shifter overflow has occurred\n");
+	if (value & ~(AIC31XX_DAC_OF_LEFT |
+		      AIC31XX_DAC_OF_RIGHT |
+		      AIC31XX_DAC_OF_SHIFTER |
+		      AIC31XX_ADC_OF |
+		      AIC31XX_ADC_OF_SHIFTER))
+		dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value);
+
+exit:
+	if (handled)
+		return IRQ_HANDLED;
+	else
+		return IRQ_NONE;
+}
+
 static int aic31xx_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
@@ -1413,6 +1477,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 	aic31xx->dev = &i2c->dev;
+	aic31xx->irq = i2c->irq;
 
 	aic31xx->codec_type = id->driver_data;
 
@@ -1456,6 +1521,26 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
+	if (aic31xx->irq > 0) {
+		regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1,
+				   AIC31XX_GPIO1_FUNC_MASK,
+				   AIC31XX_GPIO1_INT1 <<
+				   AIC31XX_GPIO1_FUNC_SHIFT);
+
+		regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL,
+			     AIC31XX_SC |
+			     AIC31XX_ENGINE);
+
+		ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq,
+						NULL, aic31xx_irq,
+						IRQF_ONESHOT, "aic31xx-irq",
+						aic31xx);
+		if (ret) {
+			dev_err(aic31xx->dev, "Unable to request IRQ\n");
+			return ret;
+		}
+	}
+
 	if (aic31xx->codec_type & DAC31XX_BIT)
 		return devm_snd_soc_register_component(&i2c->dev,
 				&soc_codec_driver_aic31xx,

sound/soc/codecs/tlv320aic31xx.h

@@ -173,6 +173,13 @@ struct aic31xx_pdata {
 #define AIC31XX_HPRDRVPWRSTATUS_MASK	BIT(1)
 #define AIC31XX_SPRDRVPWRSTATUS_MASK	BIT(0)
 
+/* AIC31XX_OFFLAG */
+#define AIC31XX_DAC_OF_LEFT		BIT(7)
+#define AIC31XX_DAC_OF_RIGHT		BIT(6)
+#define AIC31XX_DAC_OF_SHIFTER		BIT(5)
+#define AIC31XX_ADC_OF			BIT(3)
+#define AIC31XX_ADC_OF_SHIFTER		BIT(1)
+
 /* AIC31XX_INTRDACFLAG */
 #define AIC31XX_HPLSCDETECT		BIT(7)
 #define AIC31XX_HPRSCDETECT		BIT(6)
@@ -191,6 +198,22 @@ struct aic31xx_pdata {
 #define AIC31XX_SC			BIT(3)
 #define AIC31XX_ENGINE			BIT(2)
 
+/* AIC31XX_GPIO1 */
+#define AIC31XX_GPIO1_FUNC_MASK		GENMASK(5, 2)
+#define AIC31XX_GPIO1_FUNC_SHIFT	2
+#define AIC31XX_GPIO1_DISABLED		0x00
+#define AIC31XX_GPIO1_INPUT		0x01
+#define AIC31XX_GPIO1_GPI		0x02
+#define AIC31XX_GPIO1_GPO		0x03
+#define AIC31XX_GPIO1_CLKOUT		0x04
+#define AIC31XX_GPIO1_INT1		0x05
+#define AIC31XX_GPIO1_INT2		0x06
+#define AIC31XX_GPIO1_ADC_WCLK		0x07
+#define AIC31XX_GPIO1_SBCLK		0x08
+#define AIC31XX_GPIO1_SWCLK		0x09
+#define AIC31XX_GPIO1_ADC_MOD_CLK	0x10
+#define AIC31XX_GPIO1_SDOUT		0x11
+
 /* AIC31XX_DACSETUP */
 #define AIC31XX_SOFTSTEP_MASK		GENMASK(1, 0)
 

sound/soc/codecs/tscs454.c

@@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c,
 	/* Sync pg sel reg with cache */
 	regmap_write(tscs454->regmap, R_PAGESEL, 0x00);
 
-	ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
+	ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
 			tscs454_dais, ARRAY_SIZE(tscs454_dais));
 	if (ret) {
 		dev_err(&i2c->dev, "Failed to register component (%d)\n", ret);

sound/soc/codecs/wm2000.c

@@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg,
 	return regmap_write(wm2000->regmap, reg, value);
 }
 
-static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r)
-{
-	struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
-	unsigned int val;
-	int ret;
-
-	ret = regmap_read(wm2000->regmap, r, &val);
-	if (ret < 0)
-		return -1;
-
-	return val;
-}
-
 static void wm2000_reset(struct wm2000_priv *wm2000)
 {
 	struct i2c_client *i2c = wm2000->i2c;
@@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000)
 static int wm2000_poll_bit(struct i2c_client *i2c,
 			   unsigned int reg, u8 mask)
 {
+	struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
 	int timeout = 4000;
-	int val;
+	unsigned int val;
 
-	val = wm2000_read(i2c, reg);
+	regmap_read(wm2000->regmap, reg, &val);
 
 	while (!(val & mask) && --timeout) {
 		msleep(1);
-		val = wm2000_read(i2c, reg);
+		regmap_read(wm2000->regmap, reg, &val);
 	}
 
 	if (timeout == 0)
@@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 {
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 	unsigned long rate;
+	unsigned int val;
 	int ret;
 
 	if (WARN_ON(wm2000->anc_mode != ANC_OFF))
@@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 			     WM2000_MODE_THERMAL_ENABLE);
 	}
 
-	ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret);
+		regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies);
+		return ret;
+	}
 	if (wm2000->speech_clarity)
-		ret |= WM2000_SPEECH_CLARITY;
+		val |= WM2000_SPEECH_CLARITY;
 	else
-		ret &= ~WM2000_SPEECH_CLARITY;
-	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
+		val &= ~WM2000_SPEECH_CLARITY;
+	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val);
 
 	wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
 	wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02);
@@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 	const char *filename;
 	const struct firmware *fw = NULL;
 	int ret, i;
-	int reg;
+	unsigned int reg;
 	u16 id;
 
 	wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL);
@@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 	}
 
 	/* Verify that this is a WM2000 */
-	reg = wm2000_read(i2c, WM2000_REG_ID1);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret);
+		return ret;
+	}
 	id = reg << 8;
-	reg = wm2000_read(i2c, WM2000_REG_ID2);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret);
+		return ret;
+	}
 	id |= reg & 0xff;
 
 	if (id != 0x2000) {
@@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 		goto err_supplies;
 	}
 
-	reg = wm2000_read(i2c, WM2000_REG_REVISON);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret);
+		return ret;
+	}
 	dev_info(&i2c->dev, "revision %c\n", reg + 'A');
 
 	wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK");

sound/soc/codecs/wm8782.c

@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
+#include <linux/regulator/consumer.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/ac97_codec.h>
@@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = {
 	},
 };
 
+/* regulator power supply names */
+static const char *supply_names[] = {
+	"Vdda", /* analog supply, 2.7V - 3.6V */
+	"Vdd",  /* digital supply, 2.7V - 5.5V */
+};
+
+struct wm8782_priv {
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static int wm8782_soc_probe(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+static void wm8782_soc_remove(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+#ifdef CONFIG_PM
+static int wm8782_soc_suspend(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+	return 0;
+}
+
+static int wm8782_soc_resume(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+#else
+#define wm8782_soc_suspend      NULL
+#define wm8782_soc_resume       NULL
+#endif /* CONFIG_PM */
+
 static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
+	.probe			= wm8782_soc_probe,
+	.remove			= wm8782_soc_remove,
+	.suspend		= wm8782_soc_suspend,
+	.resume			= wm8782_soc_resume,
 	.dapm_widgets		= wm8782_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(wm8782_dapm_widgets),
 	.dapm_routes		= wm8782_dapm_routes,
@@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
 
 static int wm8782_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+	struct wm8782_priv *priv;
+	int ret, i;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, priv);
+
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		priv->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
+				      priv->supplies);
+	if (ret < 0)
+		return ret;
+
 	return devm_snd_soc_register_component(&pdev->dev,
 			&soc_component_dev_wm8782, &wm8782_dai, 1);
 }

sound/soc/codecs/wm8904.c

@@ -13,7 +13,6 @@
 
 #include <linux/clk.h>
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>

sound/soc/codecs/wm8974.c

@@ -11,7 +11,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>