Merge remote-tracking branch 'asoc/topic/inte' into asoc-next

Documentation/devicetree/bindings/sound/audio-graph-card.txt

@@ -10,6 +10,8 @@ see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt
 Below are same as Simple-Card.
 
 - label
+- widgets
+- routing
 - dai-format
 - frame-master
 - bitclock-master
@@ -24,6 +26,9 @@ Required properties:
 - compatible				: "audio-graph-card";
 - dais					: list of CPU DAI port{s}
 
+Optional properties:
+- pa-gpios: GPIO used to control external amplifier.
+
 Example: Single DAI case
 
 	sound_card {

Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt

@@ -90,9 +90,12 @@ Example 2. 2 CPU 1 Codec (Mixing)
 		...
 
 		port {
-			codec_endpoint: endpoint {
+			codec_endpoint0: endpoint {
 				remote-endpoint = <&cpu_endpoint0>;
 			};
+			codec_endpoint1: endpoint {
+				remote-endpoint = <&cpu_endpoint1>;
+			};
 		};
 	};
 
@@ -101,7 +104,7 @@ Example 2. 2 CPU 1 Codec (Mixing)
 		ports {
 			cpu_port0: port {
 				cpu_endpoint0: endpoint {
-					remote-endpoint = <&codec_endpoint>;
+					remote-endpoint = <&codec_endpoint0>;
 
 					dai-format = "left_j";
 					...
@@ -109,6 +112,8 @@ Example 2. 2 CPU 1 Codec (Mixing)
 			};
 			cpu_port1: port {
 				cpu_endpoint1: endpoint {
+					remote-endpoint = <&codec_endpoint1>;
+
 					dai-format = "left_j";
 					...
 				};

Documentation/devicetree/bindings/sound/nau8825.txt

@@ -69,6 +69,8 @@ Optional properties:
   - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
   - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
 
+  - nuvoton,crosstalk-bypass: make crosstalk function bypass if set.
+
   - clocks: list of phandle and clock specifier pairs according to common clock bindings for the
       clocks described in clock-names
   - clock-names: should include "mclk" for the MCLK master clock
@@ -96,6 +98,7 @@ Example:
       nuvoton,short-key-debounce = <2>;
       nuvoton,jack-insert-debounce = <7>;
       nuvoton,jack-eject-debounce = <7>;
+      nuvoton,crosstalk-bypass;
 
       clock-names = "mclk";
       clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;

Documentation/devicetree/bindings/sound/rockchip,pdm.txt

+* Rockchip PDM controller
+
+Required properties:
+
+- compatible: "rockchip,pdm"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- dmas: DMA specifiers for rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should contain following:
+   - "pdm_hclk": clock for PDM BUS
+   - "pdm_clk" : clock for PDM controller
+- pinctrl-names: Must contain a "default" entry.
+- pinctrl-N: One property must exist for each entry in
+	     pinctrl-names. See ../pinctrl/pinctrl-bindings.txt
+	     for details of the property values.
+
+Example for rk3328 PDM controller:
+
+pdm: pdm@ff040000 {
+	compatible = "rockchip,pdm";
+	reg = <0x0 0xff040000 0x0 0x1000>;
+	clocks = <&clk_pdm>, <&clk_gates28 0>;
+	clock-names = "pdm_clk", "pdm_hclk";
+	dmas = <&pdma 16>;
+	#dma-cells = <1>;
+	dma-names = "rx";
+	pinctrl-names = "default", "sleep";
+	pinctrl-0 = <&pdmm0_clk
+		     &pdmm0_fsync
+		     &pdmm0_sdi0
+		     &pdmm0_sdi1
+		     &pdmm0_sdi2
+		     &pdmm0_sdi3>;
+	pinctrl-1 = <&pdmm0_sleep>;
+	status = "disabled";
+};

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

@@ -9,7 +9,9 @@ Required properties:
 - compatible: should be one of the following:
    - "rockchip,rk3066-spdif"
    - "rockchip,rk3188-spdif"
+   - "rockchip,rk3228-spdif"
    - "rockchip,rk3288-spdif"
+   - "rockchip,rk3328-spdif"
    - "rockchip,rk3366-spdif"
    - "rockchip,rk3368-spdif"
    - "rockchip,rk3399-spdif"

Documentation/devicetree/bindings/sound/samsung,odroid.txt

@@ -5,11 +5,6 @@ Required properties:
  - compatible - "samsung,odroidxu3-audio" - for Odroid XU3 board,
 		"samsung,odroidxu4-audio" - for Odroid XU4 board
  - model - the user-visible name of this sound complex
- - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
-    controller
- - 'codec' subnode with a 'sound-dai' property containing list of phandles
-    to the CODEC nodes, first entry must be corresponding to the MAX98090
-    CODEC and the second entry must be the phandle of the HDMI IP block node
  - clocks - should contain entries matching clock names in the clock-names
     property
  - clock-names - should contain following entries:
@@ -32,12 +27,18 @@ Required properties:
    For Odroid XU4:
      no entries
 
+Required sub-nodes:
+
+ - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
+    controller
+ - 'codec' subnode with a 'sound-dai' property containing list of phandles
+    to the CODEC nodes, first entry must be corresponding to the MAX98090
+    CODEC and the second entry must be the phandle of the HDMI IP block node
+
 Example:
 
 sound {
 	compatible = "samsung,odroidxu3-audio";
-	samsung,cpu-dai = <&i2s0>;
-	samsung,codec-dai = <&max98090>;
 	model = "Odroid-XU3";
 	samsung,audio-routing =
 		"Headphone Jack", "HPL",

Documentation/devicetree/bindings/sound/st,stm32-i2s.txt

+STMicroelectronics STM32 SPI/I2S Controller
+
+The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode.
+Only some SPI instances support I2S.
+
+Required properties:
+  - compatible: Must be "st,stm32h7-i2s"
+  - reg: Offset and length of the device's register set.
+  - interrupts: Must contain the interrupt line id.
+  - clocks: Must contain phandle and clock specifier pairs for each entry
+	in clock-names.
+  - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k".
+	"i2sclk": clock which feeds the internal clock generator
+	"pclk": clock which feeds the peripheral bus interface
+	"x8k": I2S parent clock for sampling rates multiple of 8kHz.
+	"x11k": I2S parent clock for sampling rates multiple of 11.025kHz.
+  - dmas: DMA specifiers for tx and rx dma.
+    See Documentation/devicetree/bindings/dma/stm32-dma.txt.
+  - dma-names: Identifier for each DMA request line. Must be "tx" and "rx".
+  - pinctrl-names: should contain only value "default"
+  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt
+
+Optional properties:
+  - resets: Reference to a reset controller asserting the reset controller
+
+The device node should contain one 'port' child node with one child 'endpoint'
+node, according to the bindings defined in Documentation/devicetree/bindings/
+graph.txt.
+
+Example:
+sound_card {
+	compatible = "audio-graph-card";
+	dais = <&i2s2_port>;
+};
+
+i2s2: audio-controller@40003800 {
+	compatible = "st,stm32h7-i2s";
+	reg = <0x40003800 0x400>;
+	interrupts = <36>;
+	clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>;
+	clock-names = "pclk", "i2sclk",  "x8k", "x11k";
+	dmas = <&dmamux2 2 39 0x400 0x1>,
+           <&dmamux2 3 40 0x400 0x1>;
+	dma-names = "rx", "tx";
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_i2s2>;
+
+	i2s2_port: port@0 {
+		cpu_endpoint: endpoint {
+			remote-endpoint = <&codec_endpoint>;
+			format = "i2s";
+		};
+	};
+};
+
+audio-codec {
+	codec_port: port@0 {
+		codec_endpoint: endpoint {
+			remote-endpoint = <&cpu_endpoint>;
+		};
+	};
+};

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

@@ -6,7 +6,7 @@ The SAI contains two independent audio sub-blocks. Each sub-block has
 its own clock generator and I/O lines controller.
 
 Required properties:
-  - compatible: Should be "st,stm32f4-sai"
+  - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai"
   - reg: Base address and size of SAI common register set.
   - clocks: Must contain phandle and clock specifier pairs for each entry
 	in clock-names.
@@ -36,6 +36,10 @@ SAI subnodes required properties:
   - pinctrl-names: should contain only value "default"
   - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt
 
+The device node should contain one 'port' child node with one child 'endpoint'
+node, according to the bindings defined in Documentation/devicetree/bindings/
+graph.txt.
+
 Example:
 sound_card {
 	compatible = "audio-graph-card";
@@ -43,38 +47,29 @@ sound_card {
 };
 
 sai1: sai1@40015800 {
-	compatible = "st,stm32f4-sai";
+	compatible = "st,stm32h7-sai";
 	#address-cells = <1>;
 	#size-cells = <1>;
-	ranges;
+	ranges = <0 0x40015800 0x400>;
 	reg = <0x40015800 0x4>;
-	clocks = <&rcc 1 CLK_SAIQ_PDIV>, <&rcc 1 CLK_I2SQ_PDIV>;
+	clocks = <&rcc PLL1_Q>, <&rcc PLL2_P>;
 	clock-names = "x8k", "x11k";
 	interrupts = <87>;
 
-	sai1b: audio-controller@40015824 {
-		#sound-dai-cells = <0>;
-		compatible = "st,stm32-sai-sub-b";
-		reg = <0x40015824 0x1C>;
-		clocks = <&rcc 1 CLK_SAI2>;
+	sai1a: audio-controller@40015804 {
+		compatible = "st,stm32-sai-sub-a";
+		reg = <0x4 0x1C>;
+		clocks = <&rcc SAI1_CK>;
 		clock-names = "sai_ck";
-		dmas = <&dma2 5 0 0x400 0x0>;
+		dmas = <&dmamux1 1 87 0x400 0x0>;
 		dma-names = "tx";
 		pinctrl-names = "default";
-		pinctrl-0 = <&pinctrl_sai1b>;
-
-		ports {
-			#address-cells = <1>;
-			#size-cells = <0>;
+		pinctrl-0 = <&pinctrl_sai1a>;
 
-			sai1b_port: port@0 {
-				reg = <0>;
-				cpu_endpoint: endpoint {
-					remote-endpoint = <&codec_endpoint>;
-					audio-graph-card,format = "i2s";
-					audio-graph-card,bitclock-master = <&codec_endpoint>;
-					audio-graph-card,frame-master = <&codec_endpoint>;
-				};
+		sai1b_port: port {
+			cpu_endpoint: endpoint {
+				remote-endpoint = <&codec_endpoint>;
+				format = "i2s";
 			};
 		};
 	};

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

+STMicroelectronics STM32 S/PDIF receiver (SPDIFRX).
+
+The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with
+IEC-60958 and IEC-61937.
+
+Required properties:
+  - compatible: should be "st,stm32h7-spdifrx"
+  - reg: cpu DAI IP base address and size
+  - clocks: must contain an entry for kclk (used as S/PDIF signal reference)
+  - clock-names: must contain "kclk"
+  - interrupts: cpu DAI interrupt line
+  - dmas: DMA specifiers for audio data DMA and iec control flow DMA
+    See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt
+  - dma-names: two dmas have to be defined, "rx" and "rx-ctrl"
+
+Optional properties:
+  - resets: Reference to a reset controller asserting the SPDIFRX
+
+The device node should contain one 'port' child node with one child 'endpoint'
+node, according to the bindings defined in Documentation/devicetree/bindings/
+graph.txt.
+
+Example:
+spdifrx: spdifrx@40004000 {
+	compatible = "st,stm32h7-spdifrx";
+	reg = <0x40004000 0x400>;
+	clocks = <&rcc SPDIFRX_CK>;
+	clock-names = "kclk";
+	interrupts = <97>;
+	dmas = <&dmamux1 2 93 0x400 0x0>,
+	       <&dmamux1 3 94 0x400 0x0>;
+	dma-names = "rx", "rx-ctrl";
+	pinctrl-0 = <&spdifrx_pins>;
+	pinctrl-names = "default";
+
+	spdifrx_port: port {
+		cpu_endpoint: endpoint {
+			remote-endpoint = <&codec_endpoint>;
+		};
+	};
+};
+
+spdif_in: spdif-in {
+	compatible = "linux,spdif-dir";
+
+	codec_port: port {
+		codec_endpoint: endpoint {
+			remote-endpoint = <&cpu_endpoint>;
+		};
+	};
+};
+
+soundcard {
+	compatible = "audio-graph-card";
+	dais = <&spdifrx_port>;
+};

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

@@ -7,6 +7,7 @@ Required properties:
 		- "allwinner,sun7i-a20-codec"
 		- "allwinner,sun8i-a23-codec"
 		- "allwinner,sun8i-h3-codec"
+		- "allwinner,sun8i-v3s-codec"
 - reg: must contain the registers location and length
 - interrupts: must contain the codec interrupt
 - dmas: DMA channels for tx and rx dma. See the DMA client binding,
@@ -25,6 +26,7 @@ Required properties for the following compatibles:
 		- "allwinner,sun6i-a31-codec"
 		- "allwinner,sun8i-a23-codec"
 		- "allwinner,sun8i-h3-codec"
+		- "allwinner,sun8i-v3s-codec"
 - resets: phandle to the reset control for this device
 - allwinner,audio-routing: A list of the connections between audio components.
 			   Each entry is a pair of strings, the first being the
@@ -34,15 +36,15 @@ Required properties for the following compatibles:
 			   Audio pins on the SoC:
 			   "HP"
 			   "HPCOM"
-			   "LINEIN"
-			   "LINEOUT"	(not on sun8i-a23)
+			   "LINEIN"	(not on sun8i-v3s)
+			   "LINEOUT"	(not on sun8i-a23 or sun8i-v3s)
 			   "MIC1"
-			   "MIC2"
+			   "MIC2"	(not on sun8i-v3s)
 			   "MIC3"	(sun6i-a31 only)
 
 			   Microphone biases from the SoC:
 			   "HBIAS"
-			   "MBIAS"
+			   "MBIAS"	(not on sun8i-v3s)
 
 			   Board connectors:
 			   "Headphone"
@@ -55,6 +57,7 @@ Required properties for the following compatibles:
 Required properties for the following compatibles:
 		- "allwinner,sun8i-a23-codec"
 		- "allwinner,sun8i-h3-codec"
+		- "allwinner,sun8i-v3s-codec"
 - allwinner,codec-analog-controls: A phandle to the codec analog controls
 				   block in the PRCM.
 

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

@@ -4,6 +4,7 @@ Required properties:
 - compatible: must be one of the following compatibles:
 		- "allwinner,sun8i-a23-codec-analog"
 		- "allwinner,sun8i-h3-codec-analog"
+		- "allwinner,sun8i-v3s-codec-analog"
 
 Required properties if not a sub-node of the PRCM node:
 - reg: must contain the registers location and length

Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt

+ZTE ZX AUD96P22 Audio Codec
+
+Required properties:
+ - compatible: Must be "zte,zx-aud96p22"
+ - #sound-dai-cells: Should be 0
+ - reg: I2C bus slave address of AUD96P22
+
+Example:
+
+	i2c0: i2c@1486000 {
+		compatible = "zte,zx296718-i2c";
+		reg = <0x01486000 0x1000>;
+		interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>;
+		#address-cells = <1>;
+		#size-cells = <0>;
+		clocks = <&audiocrm AUDIO_I2C0_WCLK>;
+		clock-frequency = <1600000>;
+
+		aud96p22: codec@22 {
+			compatible = "zte,zx-aud96p22";
+			#sound-dai-cells = <0>;
+			reg = <0x22>;
+		};
+	};

include/sound/hdmi-codec.h

@@ -18,9 +18,11 @@
 #ifndef __HDMI_CODEC_H__
 #define __HDMI_CODEC_H__
 
+#include <linux/of_graph.h>
 #include <linux/hdmi.h>
 #include <drm/drm_edid.h>
 #include <sound/asoundef.h>
+#include <sound/soc.h>
 #include <uapi/sound/asound.h>
 
 /*
@@ -87,6 +89,13 @@ struct hdmi_codec_ops {
 	 */
 	int (*get_eld)(struct device *dev, void *data,
 		       uint8_t *buf, size_t len);
+
+	/*
+	 * Getting DAI ID
+	 * Optional
+	 */
+	int (*get_dai_id)(struct snd_soc_component *comment,
+			  struct device_node *endpoint);
 };
 
 /* HDMI codec initalization data */

include/sound/rt5645.h

@@ -21,8 +21,10 @@ struct rt5645_platform_data {
 	/* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */
 
 	unsigned int jd_mode;
-	/* Invert JD when jack insert */
-	bool jd_invert;
+	/* Use level triggered irq */
+	bool level_trigger_irq;
+	/* Invert JD1_1 status polarity */
+	bool inv_jd1_1;
 };
 
 #endif

include/sound/simple_card_utils.h

@@ -22,6 +22,11 @@ struct asoc_simple_dai {
 	struct clk *clk;
 };
 
+struct asoc_simple_card_data {
+	u32 convert_rate;
+	u32 convert_channels;
+};
+
 int asoc_simple_card_parse_daifmt(struct device *dev,
 				  struct device_node *node,
 				  struct device_node *codec,
@@ -45,6 +50,8 @@ int asoc_simple_card_parse_clk(struct device *dev,
 			       struct device_node *dai_of_node,
 			       struct asoc_simple_dai *simple_dai,
 			       const char *name);
+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)	\
@@ -73,6 +80,12 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
 				     struct device_node **endpoint_np,
 				     const char **dai_name);
 
+#define asoc_simple_card_of_parse_tdm(np, dai)			\
+	snd_soc_of_parse_tdm_slot(np,	&(dai)->tx_slot_mask,	\
+					&(dai)->rx_slot_mask,	\
+					&(dai)->slots,		\
+					&(dai)->slot_width);
+
 int asoc_simple_card_init_dai(struct snd_soc_dai *dai,
 			      struct asoc_simple_dai *simple_dai);
 
@@ -82,4 +95,15 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link,
 
 int asoc_simple_card_clean_reference(struct snd_soc_card *card);
 
+void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data,
+				      struct snd_pcm_hw_params *params);
+void asoc_simple_card_parse_convert(struct device *dev, char *prefix,
+				    struct asoc_simple_card_data *data);
+
+int asoc_simple_card_of_parse_routing(struct snd_soc_card *card,
+				      char *prefix,
+				      int optional);
+int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
+				      char *prefix);
+
 #endif /* __SIMPLE_CARD_UTILS_H */

include/sound/soc-topology.h

@@ -28,6 +28,8 @@ struct snd_soc_component;
 struct snd_soc_tplg_pcm_fe;
 struct snd_soc_dapm_context;
 struct snd_soc_card;
+struct snd_kcontrol_new;
+struct snd_soc_dai_link;
 
 /* object scan be loaded and unloaded in groups with identfying indexes */
 #define SND_SOC_TPLG_INDEX_ALL	0	/* ID that matches all FW objects */
@@ -116,6 +118,9 @@ struct snd_soc_tplg_ops {
 	int (*widget_load)(struct snd_soc_component *,
 		struct snd_soc_dapm_widget *,
 		struct snd_soc_tplg_dapm_widget *);
+	int (*widget_ready)(struct snd_soc_component *,
+		struct snd_soc_dapm_widget *,
+		struct snd_soc_tplg_dapm_widget *);
 	int (*widget_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 

sound/soc/codecs/Kconfig

@@ -1119,6 +1119,11 @@ config SND_SOC_WM9713
 	tristate
 	select REGMAP_AC97
 
+config SND_SOC_ZX_AUD96P22
+	tristate "ZTE ZX AUD96P22 CODEC"
+	depends on I2C
+	select REGMAP_I2C
+
 # Amp
 config SND_SOC_LM4857
 	tristate

sound/soc/codecs/Makefile

@@ -225,6 +225,7 @@ snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 snd-soc-wm-hubs-objs := wm_hubs.o
+snd-soc-zx-aud96p22-objs := zx_aud96p22.o
 # Amp
 snd-soc-dio2125-objs := dio2125.o
 snd-soc-max9877-objs := max9877.o
@@ -457,6 +458,7 @@ obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
 obj-$(CONFIG_SND_SOC_WM_ADSP)	+= snd-soc-wm-adsp.o
 obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
+obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o
 
 # Amp
 obj-$(CONFIG_SND_SOC_DIO2125)	+= snd-soc-dio2125.o

sound/soc/codecs/hdmi-codec.c

@@ -25,17 +25,6 @@
 
 #include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */
 
-struct hdmi_device {
-	struct device *dev;
-	struct list_head list;
-	int cnt;
-};
-#define pos_to_hdmi_device(pos)	container_of((pos), struct hdmi_device, list)
-LIST_HEAD(hdmi_device_list);
-static DEFINE_MUTEX(hdmi_mutex);
-
-#define DAI_NAME_SIZE 16
-
 #define HDMI_CODEC_CHMAP_IDX_UNKNOWN  -1
 
 struct hdmi_codec_channel_map_table {
@@ -293,7 +282,6 @@ struct hdmi_codec_priv {
 	struct hdmi_codec_daifmt daifmt[2];
 	struct mutex current_stream_lock;
 	struct snd_pcm_substream *current_stream;
-	struct snd_pcm_hw_constraint_list ratec;
 	uint8_t eld[MAX_ELD_BYTES];
 	struct snd_pcm_chmap *chmap_info;
 	unsigned int chmap_idx;
@@ -702,6 +690,7 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 }
 
 static struct snd_soc_dai_driver hdmi_i2s_dai = {
+	.name = "i2s-hifi",
 	.id = DAI_ID_I2S,
 	.playback = {
 		.stream_name = "Playback",
@@ -716,6 +705,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = {
 };
 
 static const struct snd_soc_dai_driver hdmi_spdif_dai = {
+	.name = "spdif-hifi",
 	.id = DAI_ID_SPDIF,
 	.playback = {
 		.stream_name = "Playback",
@@ -728,30 +718,16 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = {
 	.pcm_new = hdmi_codec_pcm_new,
 };
 
-static char hdmi_dai_name[][DAI_NAME_SIZE] = {
-	"hdmi-hifi.0",
-	"hdmi-hifi.1",
-	"hdmi-hifi.2",
-	"hdmi-hifi.3",
-};
-
-static int hdmi_of_xlate_dai_name(struct snd_soc_component *component,
-				  struct of_phandle_args *args,
-				  const char **dai_name)
+static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
+				 struct device_node *endpoint)
 {
-	int id;
+	struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+	int ret = -ENOTSUPP; /* see snd_soc_get_dai_id() */
 
-	if (args->args_count)
-		id = args->args[0];
-	else
-		id = 0;
+	if (hcp->hcd.ops->get_dai_id)
+		ret = hcp->hcd.ops->get_dai_id(component, endpoint);
 
-	if (id < ARRAY_SIZE(hdmi_dai_name)) {
-		*dai_name = hdmi_dai_name[id];
-		return 0;
-	}
-
-	return -EAGAIN;
+	return ret;
 }
 
 static struct snd_soc_codec_driver hdmi_codec = {
@@ -762,7 +738,7 @@ static struct snd_soc_codec_driver hdmi_codec = {
 		.num_dapm_widgets	= ARRAY_SIZE(hdmi_widgets),
 		.dapm_routes		= hdmi_routes,
 		.num_dapm_routes	= ARRAY_SIZE(hdmi_routes),
-		.of_xlate_dai_name	= hdmi_of_xlate_dai_name,
+		.of_xlate_dai_id	= hdmi_of_xlate_dai_id,
 	},
 };
 
@@ -771,8 +747,6 @@ static int hdmi_codec_probe(struct platform_device *pdev)
 	struct hdmi_codec_pdata *hcd = pdev->dev.platform_data;
 	struct device *dev = &pdev->dev;
 	struct hdmi_codec_priv *hcp;
-	struct hdmi_device *hd;
-	struct list_head *pos;
 	int dai_count, i = 0;
 	int ret;
 
@@ -794,35 +768,6 @@ static int hdmi_codec_probe(struct platform_device *pdev)
 	if (!hcp)
 		return -ENOMEM;
 
-	hd = NULL;
-	mutex_lock(&hdmi_mutex);
-	list_for_each(pos, &hdmi_device_list) {
-		struct hdmi_device *tmp = pos_to_hdmi_device(pos);
-
-		if (tmp->dev == dev->parent) {
-			hd = tmp;
-			break;
-		}
-	}
-
-	if (!hd) {
-		hd = devm_kzalloc(dev, sizeof(*hd), GFP_KERNEL);
-		if (!hd) {
-			mutex_unlock(&hdmi_mutex);
-			return -ENOMEM;
-		}
-
-		hd->dev = dev->parent;
-
-		list_add_tail(&hd->list, &hdmi_device_list);
-	}
-	mutex_unlock(&hdmi_mutex);
-
-	if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) {
-		dev_err(dev, "too many hdmi codec are deteced\n");
-		return -EINVAL;
-	}
-
 	hcp->hcd = *hcd;
 	mutex_init(&hcp->current_stream_lock);
 
@@ -835,14 +780,11 @@ static int hdmi_codec_probe(struct platform_device *pdev)
 		hcp->daidrv[i] = hdmi_i2s_dai;
 		hcp->daidrv[i].playback.channels_max =
 			hcd->max_i2s_channels;
-		hcp->daidrv[i].name = hdmi_dai_name[hd->cnt++];
 		i++;
 	}
 
-	if (hcd->spdif) {
+	if (hcd->spdif)
 		hcp->daidrv[i] = hdmi_spdif_dai;
-		hcp->daidrv[i].name = hdmi_dai_name[hd->cnt++];
-	}
 
 	ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv,
 				     dai_count);
@@ -859,20 +801,8 @@ static int hdmi_codec_probe(struct platform_device *pdev)
 static int hdmi_codec_remove(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
-	struct list_head *pos;
 	struct hdmi_codec_priv *hcp;
 
-	mutex_lock(&hdmi_mutex);
-	list_for_each(pos, &hdmi_device_list) {
-		struct hdmi_device *tmp = pos_to_hdmi_device(pos);
-
-		if (tmp->dev == dev->parent) {
-			list_del(pos);
-			break;
-		}
-	}
-	mutex_unlock(&hdmi_mutex);
-
 	hcp = dev_get_drvdata(dev);
 	kfree(hcp->chmap_info);
 	snd_soc_unregister_codec(dev);

sound/soc/codecs/max9867.c

@@ -132,7 +132,7 @@ enum rates {
 	pcm_rate_48, max_pcm_rate,
 };
 
-struct ni_div_rates {
+static const struct ni_div_rates {
 	u32 mclk;
 	u16 ni[max_pcm_rate];
 } ni_div[] = {

sound/soc/codecs/msm8916-wcd-analog.c

@@ -223,8 +223,8 @@ struct pm8916_wcd_analog_priv {
 	u16 codec_version;
 	struct clk *mclk;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
-	bool micbias1_cap_mode;
-	bool micbias2_cap_mode;
+	unsigned int micbias1_cap_mode;
+	unsigned int micbias2_cap_mode;
 };
 
 static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
@@ -285,7 +285,7 @@ static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec)
 
 static int pm8916_wcd_analog_enable_micbias_ext(struct snd_soc_codec
 						 *codec, int event,
-						 int reg, u32 cap_mode)
+						 int reg, unsigned int cap_mode)
 {
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:

sound/soc/codecs/nau8824.c

@@ -1124,6 +1124,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
+/**
+ * nau8824_set_tdm_slot - configure DAI TDM.
+ * @dai: DAI
+ * @tx_mask: Bitmask representing active TX slots. Ex.
+ *                 0xf for normal 4 channel TDM.
+ *                 0xf0 for shifted 4 channel TDM
+ * @rx_mask: Bitmask [0:1] representing active DACR RX slots.
+ *                 Bitmask [2:3] representing active DACL RX slots.
+ *                 00=CH0,01=CH1,10=CH2,11=CH3. Ex.
+ *                 0xf for DACL/R selecting TDM CH3.
+ *                 0xf0 for DACL/R selecting shifted TDM CH3.
+ * @slots: Number of slots in use.
+ * @slot_width: Width in bits for each slot.
+ *
+ * Configures a DAI for TDM operation. Only support 4 slots TDM.
+ */
+static int nau8824_set_tdm_slot(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec);
+	unsigned int tslot_l = 0, ctrl_val = 0;
+
+	if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) ||
+		((rx_mask & 0xf0) && (rx_mask & 0xf)) ||
+		((rx_mask & 0xf0) && (tx_mask & 0xf)) ||
+		((rx_mask & 0xf) && (tx_mask & 0xf0)))
+		return -EINVAL;
+
+	ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN);
+	if (tx_mask & 0xf0) {
+		tslot_l = 4 * slot_width;
+		ctrl_val |= (tx_mask >> 4);
+	} else {
+		ctrl_val |= tx_mask;
+	}
+	if (rx_mask & 0xf0)
+		ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT);
+	else
+		ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT);
+
+	regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL,
+		NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN |
+		NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK |
+		NAU8824_TDM_TX_MASK, ctrl_val);
+	regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT,
+		NAU8824_TSLOT_L_MASK, tslot_l);
+
+	return 0;
+}
+
 /**
  * nau8824_calc_fll_param - Calculate FLL parameters.
  * @fll_in: external clock provided to codec.
@@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = {
 static const struct snd_soc_dai_ops nau8824_dai_ops = {
 	.hw_params = nau8824_hw_params,
 	.set_fmt = nau8824_set_fmt,
+	.set_tdm_slot = nau8824_set_tdm_slot,
 };
 
 #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000

sound/soc/codecs/nau8824.h

@@ -258,6 +258,18 @@
 #define NAU8824_I2S_MS_SLAVE		(0 << NAU8824_I2S_MS_SFT)
 #define NAU8824_I2S_BLK_DIV_MASK	0x7
 
+/* PORT0_LEFT_TIME_SLOT (0x1E) */
+#define NAU8824_TSLOT_L_MASK	0x3ff
+
+/* TDM_CTRL (0x20) */
+#define NAU8824_TDM_MODE		(0x1 << 15)
+#define NAU8824_TDM_OFFSET_EN		(0x1 << 14)
+#define NAU8824_TDM_DACL_RX_SFT	6
+#define NAU8824_TDM_DACL_RX_MASK	(0x3 << NAU8824_TDM_DACL_RX_SFT)
+#define NAU8824_TDM_DACR_RX_SFT	4
+#define NAU8824_TDM_DACR_RX_MASK	(0x3 << NAU8824_TDM_DACR_RX_SFT)
+#define NAU8824_TDM_TX_MASK		0xf
+
 /* ADC_FILTER_CTRL (0x24) */
 #define NAU8824_ADC_SYNC_DOWN_MASK	0x3
 #define NAU8824_ADC_SYNC_DOWN_32	0

sound/soc/codecs/nau8825.c

@@ -1612,7 +1612,6 @@ static int nau8825_jack_insert(struct nau8825 *nau8825)
 		snd_soc_dapm_sync(dapm);
 		break;
 	case 2:
-	case 3:
 		dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n");
 		type = SND_JACK_HEADSET;
 
@@ -1632,6 +1631,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825)
 		snd_soc_dapm_force_enable_pin(dapm, "SAR");
 		snd_soc_dapm_sync(dapm);
 		break;
+	case 3:
+		/* detect error case */
+		dev_err(nau8825->dev, "detection error; disable mic function\n");
+		type = SND_JACK_HEADPHONE;
+		break;
 	}
 
 	/* Leaving HPOL/R grounded after jack insert by default. They will be
@@ -1682,7 +1686,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data)
 	} else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) {
 		if (nau8825_is_jack_inserted(regmap)) {
 			event |= nau8825_jack_insert(nau8825);
-			if (!nau8825->high_imped) {
+			if (!nau8825->xtalk_bypass && !nau8825->high_imped) {
 				/* Apply the cross talk suppression in the
 				 * headset without high impedance.
 				 */
@@ -2328,6 +2332,13 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_OFF:
+		/* Reset the configuration of jack type for detection */
+		/* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */
+		regmap_update_bits(nau8825->regmap, NAU8825_REG_MIC_BIAS,
+			NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0);
+		/* ground HPL/HPR, MICGRND1/2 */
+		regmap_update_bits(nau8825->regmap,
+			NAU8825_REG_HSD_CTRL, 0xf, 0xf);
 		/* Cancel and reset cross talk detection funciton */
 		nau8825_xtalk_cancel(nau8825);
 		/* Turn off all interruptions before system shutdown. Keep the
@@ -2351,6 +2362,10 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec)
 
 	disable_irq(nau8825->irq);
 	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
+	/* Power down codec power; don't suppoet button wakeup */
+	snd_soc_dapm_disable_pin(nau8825->dapm, "SAR");
+	snd_soc_dapm_disable_pin(nau8825->dapm, "MICBIAS");
+	snd_soc_dapm_sync(nau8825->dapm);
 	regcache_cache_only(nau8825->regmap, true);
 	regcache_mark_dirty(nau8825->regmap);
 
@@ -2425,10 +2440,13 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825)
 			nau8825->jack_insert_debounce);
 	dev_dbg(dev, "jack-eject-debounce:  %d\n",
 			nau8825->jack_eject_debounce);
+	dev_dbg(dev, "crosstalk-bypass:     %d\n",
+			nau8825->xtalk_bypass);
 }
 
 static int nau8825_read_device_properties(struct device *dev,
 	struct nau8825 *nau8825) {
+	int ret;
 
 	nau8825->jkdet_enable = device_property_read_bool(dev,
 		"nuvoton,jkdet-enable");
@@ -2436,30 +2454,60 @@ static int nau8825_read_device_properties(struct device *dev,
 		"nuvoton,jkdet-pull-enable");
 	nau8825->jkdet_pull_up = device_property_read_bool(dev,
 		"nuvoton,jkdet-pull-up");
-	device_property_read_u32(dev, "nuvoton,jkdet-polarity",
+	ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity",
 		&nau8825->jkdet_polarity);
-	device_property_read_u32(dev, "nuvoton,micbias-voltage",
+	if (ret)
+		nau8825->jkdet_polarity = 1;
+	ret = device_property_read_u32(dev, "nuvoton,micbias-voltage",
 		&nau8825->micbias_voltage);
-	device_property_read_u32(dev, "nuvoton,vref-impedance",
+	if (ret)
+		nau8825->micbias_voltage = 6;
+	ret = device_property_read_u32(dev, "nuvoton,vref-impedance",
 		&nau8825->vref_impedance);
-	device_property_read_u32(dev, "nuvoton,sar-threshold-num",
+	if (ret)
+		nau8825->vref_impedance = 2;
+	ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num",
 		&nau8825->sar_threshold_num);
-	device_property_read_u32_array(dev, "nuvoton,sar-threshold",
+	if (ret)
+		nau8825->sar_threshold_num = 4;
+	ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold",
 		nau8825->sar_threshold, nau8825->sar_threshold_num);
-	device_property_read_u32(dev, "nuvoton,sar-hysteresis",
+	if (ret) {
+		nau8825->sar_threshold[0] = 0x08;
+		nau8825->sar_threshold[1] = 0x12;
+		nau8825->sar_threshold[2] = 0x26;
+		nau8825->sar_threshold[3] = 0x73;
+	}
+	ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis",
 		&nau8825->sar_hysteresis);
-	device_property_read_u32(dev, "nuvoton,sar-voltage",
+	if (ret)
+		nau8825->sar_hysteresis = 0;
+	ret = device_property_read_u32(dev, "nuvoton,sar-voltage",
 		&nau8825->sar_voltage);
-	device_property_read_u32(dev, "nuvoton,sar-compare-time",
+	if (ret)
+		nau8825->sar_voltage = 6;
+	ret = device_property_read_u32(dev, "nuvoton,sar-compare-time",
 		&nau8825->sar_compare_time);
-	device_property_read_u32(dev, "nuvoton,sar-sampling-time",
+	if (ret)
+		nau8825->sar_compare_time = 1;
+	ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time",
 		&nau8825->sar_sampling_time);
-	device_property_read_u32(dev, "nuvoton,short-key-debounce",
+	if (ret)
+		nau8825->sar_sampling_time = 1;
+	ret = device_property_read_u32(dev, "nuvoton,short-key-debounce",
 		&nau8825->key_debounce);
-	device_property_read_u32(dev, "nuvoton,jack-insert-debounce",
+	if (ret)
+		nau8825->key_debounce = 3;
+	ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce",
 		&nau8825->jack_insert_debounce);
-	device_property_read_u32(dev, "nuvoton,jack-eject-debounce",
+	if (ret)
+		nau8825->jack_insert_debounce = 7;
+	ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce",
 		&nau8825->jack_eject_debounce);
+	if (ret)
+		nau8825->jack_eject_debounce = 0;
+	nau8825->xtalk_bypass = device_property_read_bool(dev,
+		"nuvoton,crosstalk-bypass");
 
 	nau8825->mclk = devm_clk_get(dev, "mclk");
 	if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) {

sound/soc/codecs/nau8825.h

@@ -476,6 +476,7 @@ struct nau8825 {
 	int xtalk_event_mask;
 	bool xtalk_protect;
 	int imp_rms[NAU8825_XTALK_IMM];
+	int xtalk_bypass;
 };
 
 int nau8825_enable_jack_detect(struct snd_soc_codec *codec,

sound/soc/codecs/rt5514.c

@@ -9,6 +9,7 @@
  * published by the Free Software Foundation.
  */
 
+#include <linux/acpi.h>
 #include <linux/fs.h>
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -395,14 +396,14 @@ static const char * const rt5514_dmic_src[] = {
 	"DMIC1", "DMIC2"
 };
 
-static const SOC_ENUM_SINGLE_DECL(
+static SOC_ENUM_SINGLE_DECL(
 	rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL,
 	RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src);
 
 static const struct snd_kcontrol_new rt5514_sto1_dmic_mux =
 	SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum);
 
-static const SOC_ENUM_SINGLE_DECL(
+static SOC_ENUM_SINGLE_DECL(
 	rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL,
 	RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src);
 
@@ -906,9 +907,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 	if (rx_mask || tx_mask)
 		val |= RT5514_TDM_MODE;
 
-	if (slots == 4)
+	switch (slots) {
+	case 4:
 		val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH;
+		break;
+
+	case 6:
+		val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH;
+		break;
 
+	case 8:
+		val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH;
+		break;
+
+	case 2:
+	default:
+		break;
+	}
 
 	switch (slot_width) {
 	case 20:
@@ -919,6 +934,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24;
 		break;
 
+	case 25:
+		val |= RT5514_TDM_MODE2;
+		break;
+
 	case 32:
 		val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32;
 		break;
@@ -930,7 +949,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 
 	regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE |
 		RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK |
-		RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val);
+		RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK |
+		RT5514_TDM_MODE2, val);
 
 	return 0;
 }
@@ -1076,6 +1096,14 @@ static const struct of_device_id rt5514_of_match[] = {
 MODULE_DEVICE_TABLE(of, rt5514_of_match);
 #endif
 
+#ifdef CONFIG_ACPI
+static struct acpi_device_id rt5514_acpi_match[] = {
+	{ "10EC5514", 0},
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match);
+#endif
+
 static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev)
 {
 	device_property_read_u32(dev, "realtek,dmic-init-delay-ms",
@@ -1179,6 +1207,7 @@ static const struct dev_pm_ops rt5514_i2_pm_ops = {
 static struct i2c_driver rt5514_i2c_driver = {
 	.driver = {
 		.name = "rt5514",
+		.acpi_match_table = ACPI_PTR(rt5514_acpi_match),
 		.of_match_table = of_match_ptr(rt5514_of_match),
 		.pm = &rt5514_i2_pm_ops,
 	},

sound/soc/codecs/rt5514.h

@@ -117,6 +117,8 @@
 #define RT5514_POW_ADCFEDL_BIT			0
 
 /* RT5514_I2S_CTRL1 (0x2010) */
+#define RT5514_TDM_MODE2			(0x1 << 30)
+#define RT5514_TDM_MODE2_SFT			30
 #define RT5514_TDM_MODE				(0x1 << 28)
 #define RT5514_TDM_MODE_SFT			28
 #define RT5514_I2S_LR_MASK			(0x1 << 26)
@@ -136,6 +138,8 @@
 #define RT5514_TDMSLOT_SEL_RX_MASK		(0x3 << 10)
 #define RT5514_TDMSLOT_SEL_RX_SFT		10
 #define RT5514_TDMSLOT_SEL_RX_4CH		(0x1 << 10)
+#define RT5514_TDMSLOT_SEL_RX_6CH		(0x2 << 10)
+#define RT5514_TDMSLOT_SEL_RX_8CH		(0x3 << 10)
 #define RT5514_CH_LEN_RX_MASK			(0x3 << 8)
 #define RT5514_CH_LEN_RX_SFT			8
 #define RT5514_CH_LEN_RX_16			(0x0 << 8)
@@ -145,6 +149,8 @@
 #define RT5514_TDMSLOT_SEL_TX_MASK		(0x3 << 6)
 #define RT5514_TDMSLOT_SEL_TX_SFT		6
 #define RT5514_TDMSLOT_SEL_TX_4CH		(0x1 << 6)
+#define RT5514_TDMSLOT_SEL_TX_6CH		(0x2 << 6)
+#define RT5514_TDMSLOT_SEL_TX_8CH		(0x3 << 6)
 #define RT5514_CH_LEN_TX_MASK			(0x3 << 4)
 #define RT5514_CH_LEN_TX_SFT			4
 #define RT5514_CH_LEN_TX_16			(0x0 << 4)

sound/soc/codecs/rt5645.c

@@ -34,6 +34,17 @@
 #include "rl6231.h"
 #include "rt5645.h"
 
+#define QUIRK_INV_JD1_1(q)	((q) & 1)
+#define QUIRK_LEVEL_IRQ(q)	(((q) >> 1) & 1)
+#define QUIRK_IN2_DIFF(q)	(((q) >> 2) & 1)
+#define QUIRK_JD_MODE(q)	(((q) >> 4) & 7)
+#define QUIRK_DMIC1_DATA_PIN(q)	(((q) >> 8) & 3)
+#define QUIRK_DMIC2_DATA_PIN(q)	(((q) >> 12) & 3)
+
+static unsigned int quirk = -1;
+module_param(quirk, uint, 0444);
+MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override");
+
 #define RT5645_DEVICE_ID 0x6308
 #define RT5650_DEVICE_ID 0x6419
 
@@ -59,7 +70,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = {
 
 static const struct reg_sequence init_list[] = {
 	{RT5645_PR_BASE + 0x3d,	0x3600},
-	{RT5645_PR_BASE + 0x1c,	0xfd20},
+	{RT5645_PR_BASE + 0x1c,	0xfd70},
 	{RT5645_PR_BASE + 0x20,	0x611f},
 	{RT5645_PR_BASE + 0x21,	0x4040},
 	{RT5645_PR_BASE + 0x23,	0x0004},
@@ -3151,7 +3162,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_sync(dapm);
 			rt5645->jack_type = SND_JACK_HEADPHONE;
 		}
-		if (rt5645->pdata.jd_invert)
+		if (rt5645->pdata.level_trigger_irq)
 			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
 				RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR);
 	} else { /* jack out */
@@ -3172,7 +3183,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_disable_pin(dapm, "LDO2");
 		snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
 		snd_soc_dapm_sync(dapm);
-		if (rt5645->pdata.jd_invert)
+		if (rt5645->pdata.level_trigger_irq)
 			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
 				RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
 	}
@@ -3238,24 +3249,16 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 		snd_soc_jack_report(rt5645->mic_jack,
 				    report, SND_JACK_MICROPHONE);
 		return;
-	case 1: /* 2 port */
-		val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070;
-		break;
-	default: /* 1 port */
-		val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020;
+	default: /* read rt5645 jd1_1 status */
+		val = snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x1000;
 		break;
 
 	}
 
-	switch (val) {
-	/* jack in */
-	case 0x30: /* 2 port */
-	case 0x0: /* 1 port or 2 port */
-		if (rt5645->jack_type == 0) {
-			report = rt5645_jack_detect(rt5645->codec, 1);
-			/* for push button and jack out */
-			break;
-		}
+	if (!val && (rt5645->jack_type == 0)) { /* jack in */
+		report = rt5645_jack_detect(rt5645->codec, 1);
+	} else if (!val && rt5645->jack_type != 0) {
+		/* for push button and jack out */
 		btn_type = 0;
 		if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) {
 			/* button pressed */
@@ -3302,19 +3305,12 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 			mod_timer(&rt5645->btn_check_timer,
 				msecs_to_jiffies(100));
 		}
-
-		break;
-	/* jack out */
-	case 0x70: /* 2 port */
-	case 0x10: /* 2 port */
-	case 0x20: /* 1 port */
+	} else {
+		/* jack out */
 		report = 0;
 		snd_soc_update_bits(rt5645->codec,
 				    RT5645_INT_IRQ_ST, 0x1, 0x0);
 		rt5645_jack_detect(rt5645->codec, 0);
-		break;
-	default:
-		break;
 	}
 
 	snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE);
@@ -3601,7 +3597,7 @@ static struct rt5645_platform_data buddy_platform_data = {
 	.dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
 	.dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
 	.jd_mode = 3,
-	.jd_invert = true,
+	.level_trigger_irq = true,
 };
 
 static struct dmi_system_id dmi_platform_intel_broadwell[] = {
@@ -3614,6 +3610,33 @@ static struct dmi_system_id dmi_platform_intel_broadwell[] = {
 	{ }
 };
 
+static struct rt5645_platform_data gpd_win_platform_data = {
+	.jd_mode = 3,
+	.inv_jd1_1 = true,
+};
+
+static const struct dmi_system_id dmi_platform_gpd_win[] = {
+	{
+		/*
+		 * Match for the GPDwin which unfortunately uses somewhat
+		 * generic dmi strings, which is why we test for 4 strings.
+		 * Comparing against 23 other byt/cht boards, board_vendor
+		 * and board_name are unique to the GPDwin, where as only one
+		 * other board has the same board_serial and 3 others have
+		 * the same default product_name. Also the GPDwin is the
+		 * only device to have both board_ and product_name not set.
+		 */
+		.ident = "GPD Win",
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
+			DMI_MATCH(DMI_BOARD_NAME, "Default string"),
+			DMI_MATCH(DMI_BOARD_SERIAL, "Default string"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Default string"),
+		},
+	},
+	{}
+};
+
 static bool rt5645_check_dp(struct device *dev)
 {
 	if (device_property_present(dev, "realtek,in2-differential") ||
@@ -3664,6 +3687,17 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 		rt5645_parse_dt(rt5645, &i2c->dev);
 	else if (dmi_check_system(dmi_platform_intel_braswell))
 		rt5645->pdata = general_platform_data;
+	else if (dmi_check_system(dmi_platform_gpd_win))
+		rt5645->pdata = gpd_win_platform_data;
+
+	if (quirk != -1) {
+		rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk);
+		rt5645->pdata.level_trigger_irq = QUIRK_LEVEL_IRQ(quirk);
+		rt5645->pdata.inv_jd1_1 = QUIRK_INV_JD1_1(quirk);
+		rt5645->pdata.jd_mode = QUIRK_JD_MODE(quirk);
+		rt5645->pdata.dmic1_data_pin = QUIRK_DMIC1_DATA_PIN(quirk);
+		rt5645->pdata.dmic2_data_pin = QUIRK_DMIC2_DATA_PIN(quirk);
+	}
 
 	rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect",
 						       GPIOD_IN);
@@ -3745,6 +3779,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 					   ret);
 	}
 
+	regmap_update_bits(rt5645->regmap, RT5645_CLSD_OUT_CTRL, 0xc0, 0xc0);
+
 	if (rt5645->pdata.in2_diff)
 		regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL,
 					RT5645_IN_DF2, RT5645_IN_DF2);
@@ -3848,12 +3884,16 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 		default:
 			break;
 		}
+		if (rt5645->pdata.inv_jd1_1) {
+			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+				RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
+		}
 	}
 
 	regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1,
 		RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2);
 
-	if (rt5645->pdata.jd_invert) {
+	if (rt5645->pdata.level_trigger_irq) {
 		regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
 			RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
 	}

sound/soc/codecs/rt5651.c

@@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control =
 	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL,
 				    RT5651_R_MUTE_SFT, 1, 1);
 
-/* INL/R source */
-static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"};
-
-static SOC_ENUM_SINGLE_DECL(
-	rt5651_inl_enum, RT5651_INL1_INR1_VOL,
-	RT5651_INL_SEL_SFT, rt5651_inl_src);
-
-static const struct snd_kcontrol_new rt5651_inl1_mux =
-	SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum);
-
-static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"};
-
-static SOC_ENUM_SINGLE_DECL(
-	rt5651_inr1_enum, RT5651_INL1_INR1_VOL,
-	RT5651_INR_SEL_SFT, rt5651_inr1_src);
-
-static const struct snd_kcontrol_new rt5651_inr1_mux =
-	SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum);
-
-static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"};
-
-static SOC_ENUM_SINGLE_DECL(
-	rt5651_inl2_enum, RT5651_INL2_INR2_VOL,
-	RT5651_INL_SEL_SFT, rt5651_inl2_src);
-
-static const struct snd_kcontrol_new rt5651_inl2_mux =
-	SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum);
-
-static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"};
-
-static SOC_ENUM_SINGLE_DECL(
-	rt5651_inr2_enum, RT5651_INL2_INR2_VOL,
-	RT5651_INR_SEL_SFT, rt5651_inr2_src);
-
-static const struct snd_kcontrol_new rt5651_inr2_mux =
-	SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum);
-
-
 /* Stereo ADC source */
 static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"};
 
@@ -955,11 +917,7 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = {
 			 RT5651_PWR_IN2_L_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL,
 			 RT5651_PWR_IN2_R_BIT, 0, NULL, 0),
-	/* IN Mux */
-	SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux),
-	SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux),
-	SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux),
-	SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux),
+
 	/* REC Mixer */
 	SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0,
 			   rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)),

sound/soc/codecs/rt5670.c

@@ -2022,7 +2022,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
 
 	{ "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" },
 	{ "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" },
-	{ "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll },
 
 	{ "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" },
 	{ "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" },
@@ -2061,7 +2060,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
 
 	{ "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" },
 	{ "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" },
-	{ "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll },
 
 	{ "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" },
 	{ "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" },
@@ -2444,10 +2442,9 @@ static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai,
-		int clk_id, unsigned int freq, int dir)
+static int rt5670_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id,
+				   int source, unsigned int freq, int dir)
 {
-	struct snd_soc_codec *codec = dai->codec;
 	struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
 	unsigned int reg_val = 0;
 
@@ -2471,7 +2468,7 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai,
 	if (clk_id != RT5670_SCLK_S_RCCLK)
 		rt5670->sysclk_src = clk_id;
 
-	dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id);
+	dev_dbg(codec->dev, "Sysclk : %dHz clock id : %d\n", freq, clk_id);
 
 	return 0;
 }
@@ -2723,7 +2720,6 @@ static int rt5670_resume(struct snd_soc_codec *codec)
 static const struct snd_soc_dai_ops rt5670_aif_dai_ops = {
 	.hw_params = rt5670_hw_params,
 	.set_fmt = rt5670_set_dai_fmt,
-	.set_sysclk = rt5670_set_dai_sysclk,
 	.set_tdm_slot = rt5670_set_tdm_slot,
 	.set_pll = rt5670_set_dai_pll,
 };
@@ -2776,6 +2772,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = {
 	.resume = rt5670_resume,
 	.set_bias_level = rt5670_set_bias_level,
 	.idle_bias_off = true,
+	.set_sysclk = rt5670_set_codec_sysclk,
 	.component_driver = {
 		.controls		= rt5670_snd_controls,
 		.num_controls		= ARRAY_SIZE(rt5670_snd_controls),

sound/soc/codecs/sgtl5000.c

@@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_DAP_AVC_DECAY,		0x0050 },
 };
 
+/* AVC: Threshold dB -> register: pre-calculated values */
+static const u16 avc_thr_db2reg[97] = {
+	0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068,
+	0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F,
+	0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414,
+	0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172,
+	0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083,
+	0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E,
+	0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010,
+	0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005,
+	0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002,
+	0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000};
+
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
 enum sgtl5000_regulator_supplies {
 	VDDA,
@@ -382,6 +396,65 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+/*
+ * custom function to get AVC threshold
+ *
+ * The threshold dB is calculated by rearranging the calculation from the
+ * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==>
+ * dB = ( fls(register_value) - 14.347 ) * 6.02
+ *
+ * As this calculation is expensive and the threshold dB values may not exeed
+ * 0 to 96 we use pre-calculated values.
+ */
+static int avc_get_threshold(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int db, i;
+	u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD);
+
+	/* register value 0 => -96dB */
+	if (!reg) {
+		ucontrol->value.integer.value[0] = 96;
+		ucontrol->value.integer.value[1] = 96;
+		return 0;
+	}
+
+	/* get dB from register value (rounded down) */
+	for (i = 0; avc_thr_db2reg[i] > reg; i++)
+		;
+	db = i;
+
+	ucontrol->value.integer.value[0] = db;
+	ucontrol->value.integer.value[1] = db;
+
+	return 0;
+}
+
+/*
+ * custom function to put AVC threshold
+ *
+ * The register value is calculated by following formula:
+ *                                    register_value = 10^(dB/20) * 0.636 * 2^15
+ * As this calculation is expensive and the threshold dB values may not exeed
+ * 0 to 96 we use pre-calculated values.
+ */
+static int avc_put_threshold(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int db;
+	u16 reg;
+
+	db = (int)ucontrol->value.integer.value[0];
+	if (db < 0 || db > 96)
+		return -EINVAL;
+	reg = avc_thr_db2reg[db];
+	snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg);
+
+	return 0;
+}
+
 static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
 
 /* tlv for mic gain, 0db 20db 30db 40db */
@@ -396,6 +469,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0);
 /* tlv for lineout volume, 31 steps of .5db each */
 static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0);
 
+/* tlv for dap avc max gain, 0db, 6db, 12db */
+static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0);
+
+/* tlv for dap avc threshold, */
+static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600);
+
 static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 	/* SOC_DOUBLE_S8_TLV with invert */
 	{
@@ -434,6 +513,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 			0x1f, 1,
 			lineout_volume),
 	SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1),
+
+	/* Automatic Volume Control (DAP AVC) */
+	SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0),
+	SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0),
+	SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0,
+			avc_max_gain),
+	SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0),
+	SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD,
+			0, 96, 0, avc_get_threshold, avc_put_threshold,
+			avc_threshold),
 };
 
 /* mute the codec used by alsa core */

sound/soc/codecs/tlv320aic31xx.c

@@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = {
 
 static struct snd_soc_dai_driver dac31xx_dai_driver[] = {
 	{
-		.name = "tlv32dac31xx-hifi",
+		.name = "tlv320dac31xx-hifi",
 		.playback = {
 			.stream_name	 = "Playback",
 			.channels_min	 = 2,

sound/soc/codecs/wm_adsp.c

@@ -482,8 +482,6 @@ struct wm_coeff_ctl_ops {
 		    struct snd_ctl_elem_value *ucontrol);
 	int (*xput)(struct snd_kcontrol *kcontrol,
 		    struct snd_ctl_elem_value *ucontrol);
-	int (*xinfo)(struct snd_kcontrol *kcontrol,
-		     struct snd_ctl_elem_info *uinfo);
 };
 
 struct wm_coeff_ctl {
@@ -1890,7 +1888,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
 	}
 
 	if (be32_to_cpu(val) != 0xbedead)
-		adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n",
+		adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n",
 			  pos + len, be32_to_cpu(val));
 
 	alg = kzalloc(len * 2, GFP_KERNEL | GFP_DMA);
@@ -2654,7 +2652,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	char preload[32];
 
-	snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift);
+	snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift);
 
 	dsp->preloaded = ucontrol->value.integer.value[0];
 

sound/soc/fsl/mpc5200_dma.c

@@ -302,7 +302,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
 	struct snd_card *card = rtd->card->snd_card;
 	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
-	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
 	size_t size = psc_dma_hardware.buffer_bytes_max;
 	int rc;
 

sound/soc/generic/audio-graph-card.c

@@ -13,6 +13,7 @@
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_device.h>
@@ -30,6 +31,34 @@ struct graph_card_data {
 		struct asoc_simple_dai codec_dai;
 	} *dai_props;
 	struct snd_soc_dai_link *dai_link;
+	struct gpio_desc *pa_gpio;
+};
+
+static int asoc_graph_card_outdrv_event(struct snd_soc_dapm_widget *w,
+					struct snd_kcontrol *kcontrol,
+					int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct graph_card_data *priv = snd_soc_card_get_drvdata(dapm->card);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		gpiod_set_value_cansleep(priv->pa_gpio, 1);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		gpiod_set_value_cansleep(priv->pa_gpio, 0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget asoc_graph_card_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV_E("Amplifier", SND_SOC_NOPM,
+			       0, 0, NULL, 0, asoc_graph_card_outdrv_event,
+			       SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 };
 
 #define graph_priv_to_card(priv) (&(priv)->snd_card)
@@ -44,13 +73,13 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream)
 	struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 	int ret;
 
-	ret = clk_prepare_enable(dai_props->cpu_dai.clk);
+	ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai);
 	if (ret)
 		return ret;
 
-	ret = clk_prepare_enable(dai_props->codec_dai.clk);
+	ret = asoc_simple_card_clk_enable(&dai_props->codec_dai);
 	if (ret)
-		clk_disable_unprepare(dai_props->cpu_dai.clk);
+		asoc_simple_card_clk_disable(&dai_props->cpu_dai);
 
 	return ret;
 }
@@ -61,9 +90,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	clk_disable_unprepare(dai_props->cpu_dai.clk);
+	asoc_simple_card_clk_disable(&dai_props->cpu_dai);
 
-	clk_disable_unprepare(dai_props->codec_dai.clk);
+	asoc_simple_card_clk_disable(&dai_props->codec_dai);
 }
 
 static struct snd_soc_ops asoc_graph_card_ops = {
@@ -100,7 +129,6 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx);
 	struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai;
 	struct asoc_simple_dai *codec_dai = &dai_props->codec_dai;
-	struct snd_soc_card *card = graph_priv_to_card(priv);
 	struct device_node *cpu_ep    = of_get_next_child(cpu_port, NULL);
 	struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep);
 	struct device_node *rcpu_ep = of_graph_get_remote_endpoint(codec_ep);
@@ -131,19 +159,11 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	ret = snd_soc_of_parse_tdm_slot(cpu_ep,
-					&cpu_dai->tx_slot_mask,
-					&cpu_dai->rx_slot_mask,
-					&cpu_dai->slots,
-					&cpu_dai->slot_width);
+	ret = asoc_simple_card_of_parse_tdm(cpu_ep, cpu_dai);
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	ret = snd_soc_of_parse_tdm_slot(codec_ep,
-					&codec_dai->tx_slot_mask,
-					&codec_dai->rx_slot_mask,
-					&codec_dai->slots,
-					&codec_dai->slot_width);
+	ret = asoc_simple_card_of_parse_tdm(codec_ep, codec_dai);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -170,7 +190,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	dai_link->init = asoc_graph_card_dai_init;
 
 	asoc_simple_card_canonicalize_cpu(dai_link,
-					  card->num_links == 1);
+		of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1);
 
 dai_link_of_err:
 	of_node_put(cpu_ep);
@@ -189,8 +209,16 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 	int rc, idx = 0;
 	int ret;
 
+	ret = asoc_simple_card_of_parse_widgets(card, NULL);
+	if (ret < 0)
+		return ret;
+
+	ret = asoc_simple_card_of_parse_routing(card, NULL, 1);
+	if (ret < 0)
+		return ret;
+
 	/*
-	 * we need to consider "widgets", "routing", "mclk-fs" around here
+	 * we need to consider "mclk-fs" around here
 	 * see simple-card
 	 */
 
@@ -242,6 +270,13 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->pa_gpio)) {
+		ret = PTR_ERR(priv->pa_gpio);
+		dev_err(dev, "failed to get amplifier gpio: %d\n", ret);
+		return ret;
+	}
+
 	priv->dai_props			= dai_props;
 	priv->dai_link			= dai_link;
 
@@ -251,6 +286,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	card->dev	= dev;
 	card->dai_link	= dai_link;
 	card->num_links	= num;
+	card->dapm_widgets = asoc_graph_card_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets);
 
 	ret = asoc_graph_card_parse_of(priv);
 	if (ret < 0) {

sound/soc/generic/audio-graph-scu-card.c

@@ -30,8 +30,7 @@ struct graph_card_data {
 	struct snd_soc_codec_conf codec_conf;
 	struct asoc_simple_dai *dai_props;
 	struct snd_soc_dai_link *dai_link;
-	u32 convert_rate;
-	u32 convert_channels;
+	struct asoc_simple_card_data adata;
 };
 
 #define graph_priv_to_card(priv) (&(priv)->snd_card)
@@ -45,7 +44,7 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream)
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	return clk_prepare_enable(dai_props->clk);
+	return asoc_simple_card_clk_enable(dai_props);
 }
 
 static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
@@ -54,7 +53,7 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	clk_disable_unprepare(dai_props->clk);
+	asoc_simple_card_clk_disable(dai_props);
 }
 
 static struct snd_soc_ops asoc_graph_card_ops = {
@@ -83,18 +82,8 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 					       struct snd_pcm_hw_params *params)
 {
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_interval *rate = hw_param_interval(params,
-						      SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
-							  SNDRV_PCM_HW_PARAM_CHANNELS);
 
-	if (priv->convert_rate)
-		rate->min =
-		rate->max = priv->convert_rate;
-
-	if (priv->convert_channels)
-		channels->min =
-		channels->max = priv->convert_channels;
+	asoc_simple_card_convert_fixup(&priv->adata, params);
 
 	return 0;
 }
@@ -136,7 +125,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 
 		/* card->num_links includes Codec */
 		asoc_simple_card_canonicalize_cpu(dai_link,
-					(card->num_links - 1) == 1);
+			of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1);
 	} else {
 		/* FE is dummy */
 		dai_link->cpu_of_node		= NULL;
@@ -167,11 +156,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 					      "prefix");
 	}
 
-	ret = snd_soc_of_parse_tdm_slot(ep,
-					&dai_props->tx_slot_mask,
-					&dai_props->rx_slot_mask,
-					&dai_props->slots,
-					&dai_props->slot_width);
+	ret = asoc_simple_card_of_parse_tdm(ep, dai_props);
 	if (ret)
 		return ret;
 
@@ -198,6 +183,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 	struct device_node *cpu_ep;
 	struct device_node *codec_ep;
 	struct device_node *rcpu_ep;
+	struct device_node *codec_port;
+	struct device_node *codec_port_old;
 	unsigned int daifmt = 0;
 	int dai_idx, ret;
 	int rc, codec;
@@ -210,15 +197,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 	 * see simple-card
 	 */
 
-	ret = snd_soc_of_parse_audio_routing(card, "routing");
-	if (ret)
+	ret = asoc_simple_card_of_parse_routing(card, NULL, 0);
+	if (ret < 0)
 		return ret;
 
-	/* sampling rate convert */
-	of_property_read_u32(node, "convert-rate", &priv->convert_rate);
-
-	/* channels transfer */
-	of_property_read_u32(node, "convert-channels", &priv->convert_channels);
+	asoc_simple_card_parse_convert(dev, NULL, &priv->adata);
 
 	/*
 	 * it supports multi CPU, single CODEC only here
@@ -254,6 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 	}
 
 	dai_idx = 0;
+	codec_port_old = NULL;
 	for (codec = 0; codec < 2; codec++) {
 		/*
 		 * To listup valid sounds continuously,
@@ -264,15 +248,22 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 			cpu_port = it.node;
 			cpu_ep   = of_get_next_child(cpu_port, NULL);
 			codec_ep = of_graph_get_remote_endpoint(cpu_ep);
+			codec_port = of_graph_get_port_parent(codec_ep);
 
 			of_node_put(cpu_port);
 			of_node_put(cpu_ep);
 			of_node_put(codec_ep);
+			of_node_put(codec_port);
 
 			if (codec) {
-				if (!codec_ep)
+				if (!codec_port)
 					continue;
 
+				if (codec_port_old == codec_port)
+					continue;
+
+				codec_port_old = codec_port;
+
 				/* Back-End (= Codec) */
 				ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0);
 				if (ret < 0)
@@ -290,9 +281,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 	if (ret)
 		goto parse_of_err;
 
-	dev_dbg(dev, "convert_rate     %d\n", priv->convert_rate);
-	dev_dbg(dev, "convert_channels %d\n", priv->convert_channels);
-
 	ret = 0;
 
 parse_of_err:
@@ -306,22 +294,34 @@ static int asoc_graph_get_dais_count(struct device *dev)
 	struct device_node *cpu_port;
 	struct device_node *cpu_ep;
 	struct device_node *codec_ep;
+	struct device_node *codec_port;
+	struct device_node *codec_port_old;
 	int count = 0;
 	int rc;
 
+	codec_port_old = NULL;
 	of_for_each_phandle(&it, rc, node, "dais", NULL, 0) {
 		cpu_port = it.node;
 		cpu_ep   = of_get_next_child(cpu_port, NULL);
 		codec_ep = of_graph_get_remote_endpoint(cpu_ep);
+		codec_port = of_graph_get_port_parent(codec_ep);
 
 		of_node_put(cpu_port);
 		of_node_put(cpu_ep);
 		of_node_put(codec_ep);
+		of_node_put(codec_port);
 
 		if (cpu_ep)
 			count++;
-		if (codec_ep)
-			count++;
+
+		if (!codec_port)
+			continue;
+
+		if (codec_port_old == codec_port)
+			continue;
+
+		count++;
+		codec_port_old = codec_port;
 	}
 
 	return count;

sound/soc/generic/simple-card-utils.c

@@ -13,6 +13,46 @@
 #include <linux/of_graph.h>
 #include <sound/simple_card_utils.h>
 
+void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data,
+				    struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	if (data->convert_rate)
+		rate->min =
+		rate->max = data->convert_rate;
+
+	if (data->convert_channels)
+		channels->min =
+		channels->max = data->convert_channels;
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_convert_fixup);
+
+void asoc_simple_card_parse_convert(struct device *dev, char *prefix,
+				    struct asoc_simple_card_data *data)
+{
+	struct device_node *np = dev->of_node;
+	char prop[128];
+
+	if (!prefix)
+		prefix = "";
+
+	/* sampling rate convert */
+	snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-rate");
+	of_property_read_u32(np, prop, &data->convert_rate);
+
+	/* channels transfer */
+	snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels");
+	of_property_read_u32(np, prop, &data->convert_channels);
+
+	dev_dbg(dev, "convert_rate     %d\n", data->convert_rate);
+	dev_dbg(dev, "convert_channels %d\n", data->convert_channels);
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert);
+
 int asoc_simple_card_parse_daifmt(struct device *dev,
 				  struct device_node *node,
 				  struct device_node *codec,
@@ -110,6 +150,24 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name);
 
+static void asoc_simple_card_clk_register(struct asoc_simple_dai *dai,
+					  struct clk *clk)
+{
+	dai->clk = clk;
+}
+
+int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai)
+{
+	return clk_prepare_enable(dai->clk);
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable);
+
+void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai)
+{
+	clk_disable_unprepare(dai->clk);
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable);
+
 int asoc_simple_card_parse_clk(struct device *dev,
 			       struct device_node *node,
 			       struct device_node *dai_of_node,
@@ -128,7 +186,8 @@ int asoc_simple_card_parse_clk(struct device *dev,
 	clk = devm_get_clk_from_child(dev, node, NULL);
 	if (!IS_ERR(clk)) {
 		simple_dai->sysclk = clk_get_rate(clk);
-		simple_dai->clk = clk;
+
+		asoc_simple_card_clk_register(simple_dai, clk);
 	} else if (!of_property_read_u32(node, "system-clock-frequency", &val)) {
 		simple_dai->sysclk = val;
 	} else {
@@ -316,6 +375,47 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card)
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference);
 
+int asoc_simple_card_of_parse_routing(struct snd_soc_card *card,
+				      char *prefix,
+				      int optional)
+{
+	struct device_node *node = card->dev->of_node;
+	char prop[128];
+
+	if (!prefix)
+		prefix = "";
+
+	snprintf(prop, sizeof(prop), "%s%s", prefix, "routing");
+
+	if (!of_property_read_bool(node, prop)) {
+		if (optional)
+			return 0;
+		return -EINVAL;
+	}
+
+	return snd_soc_of_parse_audio_routing(card, prop);
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing);
+
+int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
+				      char *prefix)
+{
+	struct device_node *node = card->dev->of_node;
+	char prop[128];
+
+	if (!prefix)
+		prefix = "";
+
+	snprintf(prop, sizeof(prop), "%s%s", prefix, "widgets");
+
+	if (of_property_read_bool(node, prop))
+		return snd_soc_of_parse_audio_simple_widgets(card, prop);
+
+	/* no widgets is not error */
+	return 0;
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets);
+
 /* Module information */
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
 MODULE_DESCRIPTION("ALSA SoC Simple Card Utils");

sound/soc/generic/simple-card.c

@@ -118,13 +118,13 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 		simple_priv_to_props(priv, rtd->num);
 	int ret;
 
-	ret = clk_prepare_enable(dai_props->cpu_dai.clk);
+	ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai);
 	if (ret)
 		return ret;
 
-	ret = clk_prepare_enable(dai_props->codec_dai.clk);
+	ret = asoc_simple_card_clk_enable(&dai_props->codec_dai);
 	if (ret)
-		clk_disable_unprepare(dai_props->cpu_dai.clk);
+		asoc_simple_card_clk_disable(&dai_props->cpu_dai);
 
 	return ret;
 }
@@ -136,9 +136,9 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
 	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	clk_disable_unprepare(dai_props->cpu_dai.clk);
+	asoc_simple_card_clk_disable(&dai_props->cpu_dai);
 
-	clk_disable_unprepare(dai_props->codec_dai.clk);
+	asoc_simple_card_clk_disable(&dai_props->codec_dai);
 }
 
 static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream,
@@ -233,13 +233,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
 	snprintf(prop, sizeof(prop), "%scpu", prefix);
 	cpu = of_get_child_by_name(node, prop);
 
+	if (!cpu) {
+		ret = -EINVAL;
+		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
+		goto dai_link_of_err;
+	}
+
 	snprintf(prop, sizeof(prop), "%splat", prefix);
 	plat = of_get_child_by_name(node, prop);
 
 	snprintf(prop, sizeof(prop), "%scodec", prefix);
 	codec = of_get_child_by_name(node, prop);
 
-	if (!cpu || !codec) {
+	if (!codec) {
 		ret = -EINVAL;
 		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
 		goto dai_link_of_err;
@@ -265,17 +271,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	ret = snd_soc_of_parse_tdm_slot(cpu,	&cpu_dai->tx_slot_mask,
-						&cpu_dai->rx_slot_mask,
-						&cpu_dai->slots,
-						&cpu_dai->slot_width);
+	ret = asoc_simple_card_of_parse_tdm(cpu, cpu_dai);
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	ret = snd_soc_of_parse_tdm_slot(codec,	&codec_dai->tx_slot_mask,
-						&codec_dai->rx_slot_mask,
-						&codec_dai->slots,
-						&codec_dai->slot_width);
+	ret = asoc_simple_card_of_parse_tdm(codec, codec_dai);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -341,12 +341,12 @@ static int asoc_simple_card_parse_aux_devs(struct device_node *node,
 	return 0;
 }
 
-static int asoc_simple_card_parse_of(struct device_node *node,
-				     struct simple_card_data *priv)
+static int asoc_simple_card_parse_of(struct simple_card_data *priv)
 {
 	struct device *dev = simple_priv_to_dev(priv);
 	struct snd_soc_card *card = simple_priv_to_card(priv);
 	struct device_node *dai_link;
+	struct device_node *node = dev->of_node;
 	int ret;
 
 	if (!node)
@@ -354,21 +354,13 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 
 	dai_link = of_get_child_by_name(node, PREFIX "dai-link");
 
-	/* The off-codec widgets */
-	if (of_property_read_bool(node, PREFIX "widgets")) {
-		ret = snd_soc_of_parse_audio_simple_widgets(card,
-					PREFIX "widgets");
-		if (ret)
-			goto card_parse_end;
-	}
+	ret = asoc_simple_card_of_parse_widgets(card, PREFIX);
+	if (ret < 0)
+		goto card_parse_end;
 
-	/* DAPM routes */
-	if (of_property_read_bool(node, PREFIX "routing")) {
-		ret = snd_soc_of_parse_audio_routing(card,
-					PREFIX "routing");
-		if (ret)
-			goto card_parse_end;
-	}
+	ret = asoc_simple_card_of_parse_routing(card, PREFIX, 1);
+	if (ret < 0)
+		goto card_parse_end;
 
 	/* Factor to mclk, used in hw_params() */
 	of_property_read_u32(node, PREFIX "mclk-fs", &priv->mclk_fs);
@@ -445,7 +437,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 
 	if (np && of_device_is_available(np)) {
 
-		ret = asoc_simple_card_parse_of(np, priv);
+		ret = asoc_simple_card_parse_of(priv);
 		if (ret < 0) {
 			if (ret != -EPROBE_DEFER)
 				dev_err(dev, "parse error %d\n", ret);

sound/soc/generic/simple-scu-card.c

@@ -27,8 +27,7 @@ struct simple_card_data {
 	struct snd_soc_codec_conf codec_conf;
 	struct asoc_simple_dai *dai_props;
 	struct snd_soc_dai_link *dai_link;
-	u32 convert_rate;
-	u32 convert_channels;
+	struct asoc_simple_card_data adata;
 };
 
 #define simple_priv_to_card(priv) (&(priv)->snd_card)
@@ -47,7 +46,7 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 	struct asoc_simple_dai *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	return clk_prepare_enable(dai_props->clk);
+	return asoc_simple_card_clk_enable(dai_props);
 }
 
 static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
@@ -57,7 +56,7 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
 	struct asoc_simple_dai *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	clk_disable_unprepare(dai_props->clk);
+	asoc_simple_card_clk_disable(dai_props);
 }
 
 static const struct snd_soc_ops asoc_simple_card_ops = {
@@ -86,18 +85,8 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 					struct snd_pcm_hw_params *params)
 {
 	struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_interval *rate = hw_param_interval(params,
-						      SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
-						SNDRV_PCM_HW_PARAM_CHANNELS);
 
-	if (priv->convert_rate)
-		rate->min =
-		rate->max = priv->convert_rate;
-
-	if (priv->convert_channels)
-		channels->min =
-		channels->max = priv->convert_channels;
+	asoc_simple_card_convert_fixup(&priv->adata, params);
 
 	return 0;
 }
@@ -171,11 +160,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 					      PREFIX "prefix");
 	}
 
-	ret = snd_soc_of_parse_tdm_slot(np,
-					&dai_props->tx_slot_mask,
-					&dai_props->rx_slot_mask,
-					&dai_props->slots,
-					&dai_props->slot_width);
+	ret = asoc_simple_card_of_parse_tdm(np, dai_props);
 	if (ret)
 		return ret;
 
@@ -206,15 +191,11 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv)
 	if (!node)
 		return -EINVAL;
 
-	ret = snd_soc_of_parse_audio_routing(card, PREFIX "routing");
+	ret = asoc_simple_card_of_parse_routing(card, PREFIX, 0);
 	if (ret < 0)
 		return ret;
 
-	/* sampling rate convert */
-	of_property_read_u32(node, PREFIX "convert-rate", &priv->convert_rate);
-
-	/* channels transfer */
-	of_property_read_u32(node, PREFIX "convert-channels", &priv->convert_channels);
+	asoc_simple_card_parse_convert(dev, PREFIX, &priv->adata);
 
 	/* find 1st codec */
 	np = of_get_child_by_name(node, PREFIX "codec");
@@ -241,9 +222,6 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv)
 	if (ret < 0)
 		return ret;
 
-	dev_dbg(dev, "convert_rate     %d\n", priv->convert_rate);
-	dev_dbg(dev, "convert_channels %d\n", priv->convert_channels);
-
 	return 0;
 }
 

sound/soc/hisilicon/hi6210-i2s.c

@@ -97,8 +97,8 @@ static inline u32 hi6210_read_reg(struct hi6210_i2s *i2s, int reg)
 	return readl(i2s->base + reg);
 }
 
-int hi6210_i2s_startup(struct snd_pcm_substream *substream,
-		     struct snd_soc_dai *cpu_dai)
+static int hi6210_i2s_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *cpu_dai)
 {
 	struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev);
 	int ret, n;
@@ -175,8 +175,9 @@ int hi6210_i2s_startup(struct snd_pcm_substream *substream,
 
 	return 0;
 }
-void hi6210_i2s_shutdown(struct snd_pcm_substream *substream,
-		       struct snd_soc_dai *cpu_dai)
+
+static void hi6210_i2s_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *cpu_dai)
 {
 	struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev);
 	int n;
@@ -524,7 +525,7 @@ static struct snd_soc_dai_ops hi6210_i2s_dai_ops = {
 	.shutdown	= hi6210_i2s_shutdown,
 };
 
-struct snd_soc_dai_driver hi6210_i2s_dai_init = {
+static const struct snd_soc_dai_driver hi6210_i2s_dai_init = {
 	.probe		= hi6210_i2s_dai_probe,
 	.playback = {
 		.channels_min = 2,

sound/soc/mediatek/mt2701/mt2701-cs42448.c

@@ -107,7 +107,7 @@ static const struct snd_kcontrol_new mt2701_cs42448_controls[] = {
 
 static const unsigned int mt2701_cs42448_sampling_rates[] = {48000};
 
-static struct snd_pcm_hw_constraint_list mt2701_cs42448_constraints_rates = {
+static const struct snd_pcm_hw_constraint_list mt2701_cs42448_constraints_rates = {
 		.count = ARRAY_SIZE(mt2701_cs42448_sampling_rates),
 		.list = mt2701_cs42448_sampling_rates,
 		.mask = 0,

sound/soc/omap/mcbsp.c

@@ -835,15 +835,11 @@ static ssize_t dma_op_mode_store(struct device *dev,
 				const char *buf, size_t size)
 {
 	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
-	const char * const *s;
-	int i = 0;
-
-	for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++)
-		if (sysfs_streq(buf, *s))
-			break;
+	int i;
 
-	if (i == ARRAY_SIZE(dma_op_modes))
-		return -EINVAL;
+	i = sysfs_match_string(dma_op_modes, buf);
+	if (i < 0)
+		return i;
 
 	spin_lock_irq(&mcbsp->lock);
 	if (!mcbsp->free) {

sound/soc/pxa/Kconfig

 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
-	depends on ARCH_PXA
+	depends on ARCH_PXA || COMPILE_TEST
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to

sound/soc/rockchip/Kconfig

@@ -15,6 +15,15 @@ config SND_SOC_ROCKCHIP_I2S
 	  Rockchip I2S device. The device supports upto maximum of
 	  8 channels each for play and record.
 
+config SND_SOC_ROCKCHIP_PDM
+	tristate "Rockchip PDM Controller Driver"
+	depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	help
+	  Say Y or M if you want to add support for PDM driver for
+	  Rockchip PDM Controller. The Controller supports up to maximum of
+	  8 channels record.
+
 config SND_SOC_ROCKCHIP_SPDIF
 	tristate "Rockchip SPDIF Device Driver"
 	depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP

sound/soc/rockchip/Makefile

 # ROCKCHIP Platform Support
 snd-soc-rockchip-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-pdm-objs := rockchip_pdm.o
 snd-soc-rockchip-spdif-objs := rockchip_spdif.o
 
 obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o
 obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
 
 snd-soc-rockchip-max98090-objs := rockchip_max98090.o

sound/soc/rockchip/rockchip_i2s.c

@@ -206,7 +206,21 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 
 	regmap_update_bits(i2s->regmap, I2S_CKR, mask, val);
 
-	mask = I2S_TXCR_IBM_MASK;
+	mask = I2S_CKR_CKP_MASK;
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		val = I2S_CKR_CKP_NEG;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		val = I2S_CKR_CKP_POS;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, I2S_CKR, mask, val);
+
+	mask = I2S_TXCR_IBM_MASK | I2S_TXCR_TFS_MASK | I2S_TXCR_PBM_MASK;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_RIGHT_J:
 		val = I2S_TXCR_IBM_RSJM;
@@ -217,13 +231,19 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_I2S:
 		val = I2S_TXCR_IBM_NORMAL;
 		break;
+	case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */
+		val = I2S_TXCR_TFS_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */
+		val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1);
+		break;
 	default:
 		return -EINVAL;
 	}
 
 	regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val);
 
-	mask = I2S_RXCR_IBM_MASK;
+	mask = I2S_RXCR_IBM_MASK | I2S_RXCR_TFS_MASK | I2S_RXCR_PBM_MASK;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_RIGHT_J:
 		val = I2S_RXCR_IBM_RSJM;
@@ -234,6 +254,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_I2S:
 		val = I2S_RXCR_IBM_NORMAL;
 		break;
+	case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */
+		val = I2S_RXCR_TFS_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */
+		val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1);
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -617,12 +643,13 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 			goto err_pm_disable;
 	}
 
-	soc_dai = devm_kzalloc(&pdev->dev,
+	soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai,
 			       sizeof(*soc_dai), GFP_KERNEL);
-	if (!soc_dai)
-		return -ENOMEM;
+	if (!soc_dai) {
+		ret = -ENOMEM;
+		goto err_pm_disable;
+	}
 
-	memcpy(soc_dai, &rockchip_i2s_dai, sizeof(*soc_dai));
 	if (!of_property_read_u32(node, "rockchip,playback-channels", &val)) {
 		if (val >= 2 && val <= 8)
 			soc_dai->playback.channels_max = val;

sound/soc/rockchip/rockchip_i2s.h

@@ -41,6 +41,7 @@
 #define I2S_TXCR_TFS_SHIFT	5
 #define I2S_TXCR_TFS_I2S	(0 << I2S_TXCR_TFS_SHIFT)
 #define I2S_TXCR_TFS_PCM	(1 << I2S_TXCR_TFS_SHIFT)
+#define I2S_TXCR_TFS_MASK	(1 << I2S_TXCR_TFS_SHIFT)
 #define I2S_TXCR_VDW_SHIFT	0
 #define I2S_TXCR_VDW(x)		((x - 1) << I2S_TXCR_VDW_SHIFT)
 #define I2S_TXCR_VDW_MASK	(0x1f << I2S_TXCR_VDW_SHIFT)
@@ -70,6 +71,7 @@
 #define I2S_RXCR_TFS_SHIFT	5
 #define I2S_RXCR_TFS_I2S	(0 << I2S_RXCR_TFS_SHIFT)
 #define I2S_RXCR_TFS_PCM	(1 << I2S_RXCR_TFS_SHIFT)
+#define I2S_RXCR_TFS_MASK	(1 << I2S_RXCR_TFS_SHIFT)
 #define I2S_RXCR_VDW_SHIFT	0
 #define I2S_RXCR_VDW(x)		((x - 1) << I2S_RXCR_VDW_SHIFT)
 #define I2S_RXCR_VDW_MASK	(0x1f << I2S_RXCR_VDW_SHIFT)
@@ -91,6 +93,7 @@
 #define I2S_CKR_CKP_SHIFT	26
 #define I2S_CKR_CKP_NEG		(0 << I2S_CKR_CKP_SHIFT)
 #define I2S_CKR_CKP_POS		(1 << I2S_CKR_CKP_SHIFT)
+#define I2S_CKR_CKP_MASK	(1 << I2S_CKR_CKP_SHIFT)
 #define I2S_CKR_RLP_SHIFT	25
 #define I2S_CKR_RLP_NORMAL	(0 << I2S_CKR_RLP_SHIFT)
 #define I2S_CKR_RLP_OPPSITE	(1 << I2S_CKR_RLP_SHIFT)

sound/soc/rockchip/rockchip_pdm.h

+/*
+ * Rockchip PDM ALSA SoC Digital Audio Interface(DAI)  driver
+ *
+ * Copyright (C) 2017 Fuzhou Rockchip Electronics Co., Ltd
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#ifndef _ROCKCHIP_PDM_H
+#define _ROCKCHIP_PDM_H
+
+/* PDM REGS */
+#define PDM_SYSCONFIG	(0x0000)
+#define PDM_CTRL0	(0x0004)
+#define PDM_CTRL1	(0x0008)
+#define PDM_CLK_CTRL	(0x000c)
+#define PDM_HPF_CTRL	(0x0010)
+#define PDM_FIFO_CTRL	(0x0014)
+#define PDM_DMA_CTRL	(0x0018)
+#define PDM_INT_EN	(0x001c)
+#define PDM_INT_CLR	(0x0020)
+#define PDM_INT_ST	(0x0024)
+#define PDM_RXFIFO_DATA	(0x0030)
+#define PDM_DATA_VALID	(0x0054)
+#define PDM_VERSION	(0x0058)
+
+/* PDM_SYSCONFIG */
+#define PDM_RX_MASK		(0x1 << 2)
+#define PDM_RX_START		(0x1 << 2)
+#define PDM_RX_STOP		(0x0 << 2)
+#define PDM_RX_CLR_MASK		(0x1 << 0)
+#define PDM_RX_CLR_WR		(0x1 << 0)
+#define PDM_RX_CLR_DONE		(0x0 << 0)
+
+/* PDM CTRL0 */
+#define PDM_PATH_MSK		(0xf << 27)
+#define PDM_PATH3_EN		BIT(30)
+#define PDM_PATH2_EN		BIT(29)
+#define PDM_PATH1_EN		BIT(28)
+#define PDM_PATH0_EN		BIT(27)
+#define PDM_HWT_EN		BIT(26)
+#define PDM_VDW_MSK		(0x1f << 0)
+#define PDM_VDW(X)		((X - 1) << 0)
+
+/* PDM CLK CTRL */
+#define PDM_CLK_MSK		BIT(5)
+#define PDM_CLK_EN		BIT(5)
+#define PDM_CLK_DIS		(0x0 << 5)
+#define PDM_CKP_MSK		BIT(3)
+#define PDM_CKP_NORMAL		(0x0 << 3)
+#define PDM_CKP_INVERTED	BIT(3)
+#define PDM_DS_RATIO_MSK	(0x7 << 0)
+#define PDM_CLK_320FS		(0x0 << 0)
+#define PDM_CLK_640FS		(0x1 << 0)
+#define PDM_CLK_1280FS		(0x2 << 0)
+#define PDM_CLK_2560FS		(0x3 << 0)
+#define PDM_CLK_5120FS		(0x4 << 0)
+
+/* PDM HPF CTRL */
+#define PDM_HPF_LE		BIT(3)
+#define PDM_HPF_RE		BIT(2)
+#define PDM_HPF_CF_MSK		(0x3 << 0)
+#define PDM_HPF_3P79HZ		(0x0 << 0)
+#define PDM_HPF_60HZ		(0x1 << 0)
+#define PDM_HPF_243HZ		(0x2 << 0)
+#define PDM_HPF_493HZ		(0x3 << 0)
+
+/* PDM DMA CTRL */
+#define PDM_DMA_RD_MSK		BIT(8)
+#define PDM_DMA_RD_EN		BIT(8)
+#define PDM_DMA_RD_DIS		(0x0 << 8)
+#define PDM_DMA_RDL_MSK		(0x7f << 0)
+#define PDM_DMA_RDL(X)		((X - 1) << 0)
+
+#endif /* _ROCKCHIP_PDM_H */

sound/soc/rockchip/rockchip_spdif.c

@@ -49,8 +49,12 @@ static const struct of_device_id rk_spdif_match[] = {
 	  .data = (void *)RK_SPDIF_RK3066 },
 	{ .compatible = "rockchip,rk3188-spdif",
 	  .data = (void *)RK_SPDIF_RK3188 },
+	{ .compatible = "rockchip,rk3228-spdif",
+	  .data = (void *)RK_SPDIF_RK3366 },
 	{ .compatible = "rockchip,rk3288-spdif",
 	  .data = (void *)RK_SPDIF_RK3288 },
+	{ .compatible = "rockchip,rk3328-spdif",
+	  .data = (void *)RK_SPDIF_RK3366 },
 	{ .compatible = "rockchip,rk3366-spdif",
 	  .data = (void *)RK_SPDIF_RK3366 },
 	{ .compatible = "rockchip,rk3368-spdif",

sound/soc/samsung/s3c24xx_uda134x.c

@@ -44,7 +44,7 @@ struct s3c24xx_uda134x {
 
 static unsigned int rates[33 * 2];
 #ifdef ENFORCE_RATES
-static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+static const struct snd_pcm_hw_constraint_list hw_constraints_rates = {
 	.count	= ARRAY_SIZE(rates),
 	.list	= rates,
 	.mask	= 0,

sound/soc/sh/fsi.c

@@ -301,7 +301,12 @@ struct fsi_master {
 	spinlock_t lock;
 };
 
-static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io);
+static inline int fsi_stream_is_play(struct fsi_priv *fsi,
+				     struct fsi_stream *io)
+{
+	return &fsi->playback == io;
+}
+
 
 /*
  *		basic read write function
@@ -489,12 +494,6 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi)
 /*
  *		fsi_stream_xx() function
  */
-static inline int fsi_stream_is_play(struct fsi_priv *fsi,
-				     struct fsi_stream *io)
-{
-	return &fsi->playback == io;
-}
-
 static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi,
 					struct snd_pcm_substream *substream)
 {

sound/soc/soc-topology.c

@@ -352,6 +352,17 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg,
 	return 0;
 }
 
+/* optionally pass new dynamic widget to component driver. This is mainly for
+ * external widgets where we can assign private data/ops */
+static int soc_tplg_widget_ready(struct soc_tplg *tplg,
+	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
+{
+	if (tplg->comp && tplg->ops && tplg->ops->widget_ready)
+		return tplg->ops->widget_ready(tplg->comp, w, tplg_w);
+
+	return 0;
+}
+
 /* pass DAI configurations to component driver for extra initialization */
 static int soc_tplg_dai_load(struct soc_tplg *tplg,
 	struct snd_soc_dai_driver *dai_drv)
@@ -1160,7 +1171,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		return -EINVAL;
 	}
 
-	dev_dbg(tplg->dev, "ASoC: adding %d DAPM routes\n", count);
+	dev_dbg(tplg->dev, "ASoC: adding %d DAPM routes for index %d\n", count,
+		hdr->index);
 
 	for (i = 0; i < count; i++) {
 		elem = (struct snd_soc_tplg_dapm_graph_elem *)tplg->pos;
@@ -1473,6 +1485,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg,
 	if (template.id < 0)
 		return template.id;
 
+	/* strings are allocated here, but used and freed by the widget */
 	template.name = kstrdup(w->name, GFP_KERNEL);
 	if (!template.name)
 		return -ENOMEM;
@@ -1585,11 +1598,17 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg,
 	widget->dobj.widget.kcontrol_type = kcontrol_type;
 	widget->dobj.ops = tplg->ops;
 	widget->dobj.index = tplg->index;
-	kfree(template.sname);
-	kfree(template.name);
 	list_add(&widget->dobj.list, &tplg->comp->dobj_list);
+
+	ret = soc_tplg_widget_ready(tplg, widget, w);
+	if (ret < 0)
+		goto ready_err;
+
 	return 0;
 
+ready_err:
+	snd_soc_tplg_widget_remove(widget);
+	snd_soc_dapm_free_widget(widget);
 hdr_err:
 	kfree(template.sname);
 err:
@@ -1636,7 +1655,7 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg)
 	*/
 	if (!card || !card->instantiated) {
 		dev_warn(tplg->dev, "ASoC: Parent card not yet available,"
-				"Do not add new widgets now\n");
+			" widget card binding deferred\n");
 		return 0;
 	}
 
@@ -2371,7 +2390,7 @@ static int soc_tplg_load_header(struct soc_tplg *tplg,
 
 	/* check for matching ID */
 	if (hdr->index != tplg->req_index &&
-		hdr->index != SND_SOC_TPLG_INDEX_ALL)
+		tplg->req_index != SND_SOC_TPLG_INDEX_ALL)
 		return 0;
 
 	tplg->index = hdr->index;

sound/soc/stm/Kconfig

-menuconfig SND_SOC_STM32
-	tristate "STMicroelectronics STM32 SOC audio support"
+menu "STMicroelectronics STM32 SOC audio support"
+
+config SND_SOC_STM32_SAI
+	tristate "STM32 SAI interface (Serial Audio Interface) support"
 	depends on ARCH_STM32 || COMPILE_TEST
 	depends on SND_SOC
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select REGMAP_MMIO
 	help
-	  Say Y if you want to enable ASoC-support for STM32
+	  Say Y if you want to enable SAI for STM32
+
+config SND_SOC_STM32_I2S
+	tristate "STM32 I2S interface (SPI/I2S block) support"
+	depends on ARCH_STM32 || COMPILE_TEST
+	depends on SND_SOC
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  Say Y if you want to enable I2S for STM32
+
+config SND_SOC_STM32_SPDIFRX
+	tristate "STM32 S/PDIF receiver (SPDIFRX) support"
+	depends on ARCH_STM32 || COMPILE_TEST
+	depends on SND_SOC
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	select SND_SOC_SPDIF
+	help
+	  Say Y if you want to enable S/PDIF capture for STM32
+
+endmenu

sound/soc/stm/Makefile

 # SAI
 snd-soc-stm32-sai-sub-objs := stm32_sai_sub.o
-obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai-sub.o
+obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai-sub.o
 
 snd-soc-stm32-sai-objs := stm32_sai.o
-obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai.o
+obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai.o
+
+# I2S
+snd-soc-stm32-i2s-objs := stm32_i2s.o
+obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o
+
+# SPDIFRX
+snd-soc-stm32-spdifrx-objs := stm32_spdifrx.o
+obj-$(CONFIG_SND_SOC_STM32_SPDIFRX) += snd-soc-stm32-spdifrx.o

sound/soc/stm/stm32_sai.c

@@ -27,8 +27,17 @@
 
 #include "stm32_sai.h"
 
+static const struct stm32_sai_conf stm32_sai_conf_f4 = {
+	.version = SAI_STM32F4,
+};
+
+static const struct stm32_sai_conf stm32_sai_conf_h7 = {
+	.version = SAI_STM32H7,
+};
+
 static const struct of_device_id stm32_sai_ids[] = {
-	{ .compatible = "st,stm32f4-sai", .data = (void *)SAI_STM32F4 },
+	{ .compatible = "st,stm32f4-sai", .data = (void *)&stm32_sai_conf_f4 },
+	{ .compatible = "st,stm32h7-sai", .data = (void *)&stm32_sai_conf_h7 },
 	{}
 };
 
@@ -52,7 +61,7 @@ static int stm32_sai_probe(struct platform_device *pdev)
 
 	of_id = of_match_device(stm32_sai_ids, &pdev->dev);
 	if (of_id)
-		sai->version = (enum stm32_sai_version)of_id->data;
+		sai->conf = (struct stm32_sai_conf *)of_id->data;
 	else
 		return -EINVAL;
 
@@ -110,6 +119,6 @@ static struct platform_driver stm32_sai_driver = {
 module_platform_driver(stm32_sai_driver);
 
 MODULE_DESCRIPTION("STM32 Soc SAI Interface");
-MODULE_AUTHOR("Olivier Moysan, <olivier.moysan@st.com>");
+MODULE_AUTHOR("Olivier Moysan <olivier.moysan@st.com>");
 MODULE_ALIAS("platform:st,stm32-sai");
 MODULE_LICENSE("GPL v2");