Merge branch 'asoc-4.22' into asoc-5.0

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

-Audio-Graph-SCU-Card:
-
-Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM".
-
-It is based on common bindings for device graphs.
-see ${LINUX}/Documentation/devicetree/bindings/graph.txt
-
-Basically, Audio-Graph-SCU-Card property is same as
-Simple-Card / Simple-SCU-Card / Audio-Graph-Card.
-see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt
-    ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt
-    ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt
-
-Below are same as Simple-Card / Audio-Graph-Card.
-
-- label
-- dai-format
-- frame-master
-- bitclock-master
-- bitclock-inversion
-- frame-inversion
-- dai-tdm-slot-num
-- dai-tdm-slot-width
-- clocks / system-clock-frequency
-
-Below are same as Simple-SCU-Card.
-
-- convert-rate
-- convert-channels
-- prefix
-- routing
-
-Required properties:
-
-- compatible				: "audio-graph-scu-card";
-- dais					: list of CPU DAI port{s}
-
-Example 1. Sampling Rate Conversion
-
-	sound_card {
-		compatible = "audio-graph-scu-card";
-
-		label = "sound-card";
-		prefix = "codec";
-		routing = "codec Playback", "DAI0 Playback",
-			  "DAI0 Capture",   "codec Capture";
-		convert-rate = <48000>;
-
-		dais = <&cpu_port>;
-	};
-
-	audio-codec {
-		...
-
-		port {
-			codec_endpoint: endpoint {
-				remote-endpoint = <&cpu_endpoint>;
-			};
-		};
-	};
-
-	dai-controller {
-		...
-		cpu_port: port {
-			cpu_endpoint: endpoint {
-				remote-endpoint = <&codec_endpoint>;
-
-				dai-format = "left_j";
-				...
-			};
-		};
-	};
-
-Example 2. 2 CPU 1 Codec (Mixing)
-
-	sound_card {
-		compatible = "audio-graph-scu-card";
-
-		label = "sound-card";
-		routing = "codec Playback", "DAI0 Playback",
-			  "codec Playback", "DAI1 Playback",
-			  "DAI0 Capture",   "codec Capture";
-
-		dais = <&cpu_port0
-			&cpu_port1>;
-	};
-
-	audio-codec {
-		...
-
-		audio-graph-card,prefix = "codec";
-		audio-graph-card,convert-rate = <48000>;
-		port {
-			codec_endpoint0: endpoint {
-				remote-endpoint = <&cpu_endpoint0>;
-			};
-			codec_endpoint1: endpoint {
-				remote-endpoint = <&cpu_endpoint1>;
-			};
-		};
-	};
-
-	dai-controller {
-		...
-		ports {
-			cpu_port0: port {
-				cpu_endpoint0: endpoint {
-					remote-endpoint = <&codec_endpoint0>;
-
-					dai-format = "left_j";
-					...
-				};
-			};
-			cpu_port1: port {
-				cpu_endpoint1: endpoint {
-					remote-endpoint = <&codec_endpoint1>;
-
-					dai-format = "left_j";
-					...
-				};
-			};
-		};
-	};

Documentation/devicetree/bindings/sound/cs4341.txt

+Cirrus Logic CS4341 audio DAC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+  - compatible: "cirrus,cs4341a"
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+For required properties on I2C-bus, please consult
+Documentation/devicetree/bindings/i2c/i2c.txt
+For required properties on SPI-bus, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Example:
+	codec: cs4341@0 {
+		#sound-dai-cells = <0>;
+		compatible = "cirrus,cs4341a";
+		reg = <0>;
+		spi-max-frequency = <6000000>;
+	};

Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt

@@ -45,6 +45,23 @@ Optional properties:
 - fck_parent : Should contain a valid clock name which will be used as parent
 	       for the McASP fck
 
+Optional GPIO support:
+If any McASP pin need to be used as GPIO then the McASP node must have:
+...
+  gpio-controller
+  #gpio-cells = <2>;
+...
+
+When requesting a GPIO, the first parameter is the PIN index in McASP_P*
+registers.
+For example to request the AXR2 pin of mcasp8:
+function-gpios = <&mcasp8 2 0>;
+
+Or to request the ACLKR pin of mcasp8:
+function-gpios = <&mcasp8 29 0>;
+
+For generic gpio information, please refer to bindings/gpio/gpio.txt
+
 Example:
 
 mcasp0: mcasp0@1d00000 {

Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt

+* Rockchip Rk3328 internal codec
+
+Required properties:
+
+- compatible: "rockchip,rk3328-codec"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- rockchip,grf: the phandle of the syscon node for GRF register.
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should be "pclk".
+- spk-depop-time-ms: speak depop time msec.
+
+Example for rk3328 internal codec:
+
+codec: codec@ff410000 {
+	compatible = "rockchip,rk3328-codec";
+	reg = <0x0 0xff410000 0x0 0x1000>;
+	rockchip,grf = <&grf>;
+	clocks = <&cru PCLK_ACODEC>;
+	clock-names = "pclk";
+	spk-depop-time-ms = 100;
+	status = "disabled";
+};

Documentation/devicetree/bindings/sound/simple-scu-card.txt

-ASoC Simple SCU Sound Card
-
-Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM".
-For example, you can use this driver if you want to exchange sampling rate convert,
-Mixing, etc...
-
-Required properties:
-
-- compatible				: "simple-scu-audio-card"
-					  "renesas,rsrc-card"
-Optional properties:
-
-- simple-audio-card,name		: see simple-audio-card.txt
-- simple-audio-card,cpu			: see simple-audio-card.txt
-- simple-audio-card,codec		: see simple-audio-card.txt
-
-Optional subnode properties:
-
-- simple-audio-card,format		: see simple-audio-card.txt
-- simple-audio-card,frame-master	: see simple-audio-card.txt
-- simple-audio-card,bitclock-master	: see simple-audio-card.txt
-- simple-audio-card,bitclock-inversion	: see simple-audio-card.txt
-- simple-audio-card,frame-inversion	: see simple-audio-card.txt
-- simple-audio-card,convert-rate	: platform specified sampling rate convert
-- simple-audio-card,convert-channels	: platform specified converted channel size (2 - 8 ch)
-- simple-audio-card,prefix		: see routing
-- simple-audio-card,widgets		: Please refer to widgets.txt.
-- simple-audio-card,routing		: A list of the connections between audio components.
-					  Each entry is a pair of strings, the first being the connection's sink,
-					  the second being the connection's source. Valid names for sources.
-					  use audio-prefix if some components is using same sink/sources naming.
-					  it can be used if compatible was "renesas,rsrc-card";
-
-Required CPU/CODEC subnodes properties:
-
-- sound-dai				: see simple-audio-card.txt
-
-Optional CPU/CODEC subnodes properties:
-
-- clocks / system-clock-frequency	: see simple-audio-card.txt
-
-Example 1. Sampling Rate Conversion
-
-sound {
-	compatible = "simple-scu-audio-card";
-
-	simple-audio-card,name = "rsnd-ak4643";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&sndcodec>;
-	simple-audio-card,frame-master = <&sndcodec>;
-
-	simple-audio-card,convert-rate = <48000>;
-
-	simple-audio-card,prefix = "ak4642";
-	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
-			"DAI0 Capture", "ak4642 Capture";
-
-	sndcpu: simple-audio-card,cpu {
-		sound-dai = <&rcar_sound>;
-	};
-
-	sndcodec: simple-audio-card,codec {
-		sound-dai = <&ak4643>;
-		system-clock-frequency = <11289600>;
-	};
-};
-
-Example 2. 2 CPU 1 Codec (Mixing)
-
-sound {
-	compatible = "simple-scu-audio-card";
-
-	simple-audio-card,name = "rsnd-ak4643";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&dpcmcpu>;
-	simple-audio-card,frame-master = <&dpcmcpu>;
-
-	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
-			"ak4642 Playback", "DAI1 Playback";
-
-	dpcmcpu: cpu@0 {
-		sound-dai = <&rcar_sound 0>;
-	};
-
-	cpu@1 {
-		sound-dai = <&rcar_sound 1>;
-	};
-
-	codec {
-		prefix = "ak4642";
-		sound-dai = <&ak4643>;
-		clocks = <&audio_clock>;
-	};
-};

Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt

+Device-Tree bindings for Xilinx PL audio formatter
+
+The IP core supports DMA, data formatting(AES<->PCM conversion)
+of audio samples.
+
+Required properties:
+ - compatible: "xlnx,audio-formatter-1.0"
+ - interrupt-names: Names specified to list of interrupts in same
+		    order mentioned under "interrupts".
+		    List of supported interrupt names are:
+		    "irq_mm2s" : interrupt from MM2S block
+		    "irq_s2mm" : interrupt from S2MM block
+ - interrupts-parent: Phandle for interrupt controller.
+ - interrupts: List of Interrupt numbers.
+ - reg: Base address and size of the IP core instance.
+ - clock-names: List of input clocks.
+   Required elements: "s_axi_lite_aclk", "aud_mclk"
+ - clocks: Input clock specifier. Refer to common clock bindings.
+
+Example:
+	audio_ss_0_audio_formatter_0: audio_formatter@80010000 {
+		compatible = "xlnx,audio-formatter-1.0";
+		interrupt-names = "irq_mm2s", "irq_s2mm";
+		interrupt-parent = <&gic>;
+		interrupts = <0 104 4>, <0 105 4>;
+		reg = <0x0 0x80010000 0x0 0x1000>;
+		clock-names = "s_axi_lite_aclk", "aud_mclk";
+		clocks = <&clk 71>, <&clk_wiz_1 0>;
+	};

sound/core/compress_offload.c

@@ -541,7 +541,8 @@ static int snd_compress_check_input(struct snd_compr_params *params)
 {
 	/* first let's check the buffer parameter's */
 	if (params->buffer.fragment_size == 0 ||
-	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size)
+	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size ||
+	    params->buffer.fragments == 0)
 		return -EINVAL;
 
 	/* now codec parameters */

sound/soc/codecs/Kconfig

@@ -65,6 +65,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS4271_SPI if SPI_MASTER
 	select SND_SOC_CS42XX8_I2C if I2C
 	select SND_SOC_CS43130 if I2C
+	select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CS4349 if I2C
 	select SND_SOC_CS47L24 if MFD_CS47L24
 	select SND_SOC_CS53L30 if I2C
@@ -129,6 +130,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM5102A
 	select SND_SOC_PCM512x_I2C if I2C
 	select SND_SOC_PCM512x_SPI if SPI_MASTER
+	select SND_SOC_RK3328
 	select SND_SOC_RT274 if I2C
 	select SND_SOC_RT286 if I2C
 	select SND_SOC_RT298 if I2C
@@ -542,6 +544,12 @@ config SND_SOC_CS43130
         tristate "Cirrus Logic CS43130 CODEC"
         depends on I2C
 
+config SND_SOC_CS4341
+	tristate "Cirrus Logic CS4341 CODEC"
+	depends on I2C || SPI_MASTER
+	select REGMAP_I2C if I2C
+	select REGMAP_SPI if SPI_MASTER
+
 # Cirrus Logic CS4349 HiFi DAC
 config SND_SOC_CS4349
 	tristate "Cirrus Logic CS4349 CODEC"
@@ -799,6 +807,10 @@ config SND_SOC_PCM512x_SPI
 	select SND_SOC_PCM512x
 	select REGMAP_SPI
 
+config SND_SOC_RK3328
+	tristate "Rockchip RK3328 audio CODEC"
+	select REGMAP_MMIO
+
 config SND_SOC_RL6231
 	tristate
 	default y if SND_SOC_RT5514=y
@@ -1211,7 +1223,8 @@ config SND_SOC_WM8903
 	depends on I2C
 
 config SND_SOC_WM8904
-	tristate
+	tristate "Wolfson Microelectronics WM8904 CODEC"
+	depends on I2C
 
 config SND_SOC_WM8940
         tristate

sound/soc/codecs/Makefile

@@ -60,6 +60,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o
 snd-soc-cs42xx8-objs := cs42xx8.o
 snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
 snd-soc-cs43130-objs := cs43130.o
+snd-soc-cs4341-objs := cs4341.o
 snd-soc-cs4349-objs := cs4349.o
 snd-soc-cs47l24-objs := cs47l24.o
 snd-soc-cs53l30-objs := cs53l30.o
@@ -132,6 +133,7 @@ snd-soc-pcm5102a-objs := pcm5102a.o
 snd-soc-pcm512x-objs := pcm512x.o
 snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
 snd-soc-pcm512x-spi-objs := pcm512x-spi.o
+snd-soc-rk3328-objs := rk3328_codec.o
 snd-soc-rl6231-objs := rl6231.o
 snd-soc-rl6347a-objs := rl6347a.o
 snd-soc-rt1305-objs := rt1305.o
@@ -326,6 +328,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI)	+= snd-soc-cs4271-spi.o
 obj-$(CONFIG_SND_SOC_CS42XX8)	+= snd-soc-cs42xx8.o
 obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
 obj-$(CONFIG_SND_SOC_CS43130)   += snd-soc-cs43130.o
+obj-$(CONFIG_SND_SOC_CS4341)	+= snd-soc-cs4341.o
 obj-$(CONFIG_SND_SOC_CS4349)	+= snd-soc-cs4349.o
 obj-$(CONFIG_SND_SOC_CS47L24)	+= snd-soc-cs47l24.o
 obj-$(CONFIG_SND_SOC_CS53L30)	+= snd-soc-cs53l30.o
@@ -398,6 +401,7 @@ obj-$(CONFIG_SND_SOC_PCM5102A)	+= snd-soc-pcm5102a.o
 obj-$(CONFIG_SND_SOC_PCM512x)	+= snd-soc-pcm512x.o
 obj-$(CONFIG_SND_SOC_PCM512x_I2C)	+= snd-soc-pcm512x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
+obj-$(CONFIG_SND_SOC_RK3328)	+= snd-soc-rk3328.o
 obj-$(CONFIG_SND_SOC_RL6231)	+= snd-soc-rl6231.o
 obj-$(CONFIG_SND_SOC_RL6347A)	+= snd-soc-rl6347a.o
 obj-$(CONFIG_SND_SOC_RT1305)	+= snd-soc-rt1305.o

sound/soc/codecs/cs4341.c

+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ *  Cirrus Logic CS4341A ALSA SoC Codec Driver
+ *  Author: Alexander Shiyan <shc_work@mail.ru>
+ */
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#define CS4341_REG_MODE1	0x00
+#define CS4341_REG_MODE2	0x01
+#define CS4341_REG_MIX		0x02
+#define CS4341_REG_VOLA		0x03
+#define CS4341_REG_VOLB		0x04
+
+#define CS4341_MODE2_DIF	(7 << 4)
+#define CS4341_MODE2_DIF_I2S_24	(0 << 4)
+#define CS4341_MODE2_DIF_I2S_16	(1 << 4)
+#define CS4341_MODE2_DIF_LJ_24	(2 << 4)
+#define CS4341_MODE2_DIF_RJ_24	(3 << 4)
+#define CS4341_MODE2_DIF_RJ_16	(5 << 4)
+#define CS4341_VOLX_MUTE	(1 << 7)
+
+struct cs4341_priv {
+	unsigned int		fmt;
+	struct regmap		*regmap;
+	struct regmap_config	regcfg;
+};
+
+static const struct reg_default cs4341_reg_defaults[] = {
+	{ CS4341_REG_MODE1,	0x00 },
+	{ CS4341_REG_MODE2,	0x82 },
+	{ CS4341_REG_MIX,	0x49 },
+	{ CS4341_REG_VOLA,	0x80 },
+	{ CS4341_REG_VOLB,	0x80 },
+};
+
+static int cs4341_set_fmt(struct snd_soc_dai *dai, unsigned int format)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs4341_priv *cs4341 = snd_soc_component_get_drvdata(component);
+
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (format & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		cs4341->fmt = format & SND_SOC_DAIFMT_FORMAT_MASK;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int cs4341_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs4341_priv *cs4341 = snd_soc_component_get_drvdata(component);
+	unsigned int mode = 0;
+	int b24 = 0;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S24_LE:
+		b24 = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	default:
+		dev_err(component->dev, "Unsupported PCM format 0x%08x.\n",
+			params_format(params));
+		return -EINVAL;
+	}
+
+	switch (cs4341->fmt) {
+	case SND_SOC_DAIFMT_I2S:
+		mode = b24 ? CS4341_MODE2_DIF_I2S_24 : CS4341_MODE2_DIF_I2S_16;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode = CS4341_MODE2_DIF_LJ_24;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode = b24 ? CS4341_MODE2_DIF_RJ_24 : CS4341_MODE2_DIF_RJ_16;
+		break;
+	default:
+		dev_err(component->dev, "Unsupported DAI format 0x%08x.\n",
+			cs4341->fmt);
+		return -EINVAL;
+	}
+
+	return snd_soc_component_update_bits(component, CS4341_REG_MODE2,
+					     CS4341_MODE2_DIF, mode);
+}
+
+static int cs4341_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_component *component = dai->component;
+	int ret;
+
+	ret = snd_soc_component_update_bits(component, CS4341_REG_VOLA,
+					    CS4341_VOLX_MUTE,
+					    mute ? CS4341_VOLX_MUTE : 0);
+	if (ret < 0)
+		return ret;
+
+	return snd_soc_component_update_bits(component, CS4341_REG_VOLB,
+					     CS4341_VOLX_MUTE,
+					     mute ? CS4341_VOLX_MUTE : 0);
+}
+
+static DECLARE_TLV_DB_SCALE(out_tlv, -9000, 100, 0);
+
+static const char * const deemph[] = {
+	"None", "44.1k", "48k", "32k",
+};
+
+static const struct soc_enum deemph_enum =
+	SOC_ENUM_SINGLE(CS4341_REG_MODE2, 2, 4, deemph);
+
+static const char * const srzc[] = {
+	"Immediate", "Zero Cross", "Soft Ramp", "SR on ZC",
+};
+
+static const struct soc_enum srzc_enum =
+	SOC_ENUM_SINGLE(CS4341_REG_MIX, 5, 4, srzc);
+
+
+static const struct snd_soc_dapm_widget cs4341_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUTPUT("OutA"),
+	SND_SOC_DAPM_OUTPUT("OutB"),
+};
+
+static const struct snd_soc_dapm_route cs4341_routes[] = {
+	{ "OutA", NULL, "HiFi DAC" },
+	{ "OutB", NULL, "HiFi DAC" },
+	{ "DAC Playback", NULL, "OutA" },
+	{ "DAC Playback", NULL, "OutB" },
+};
+
+static const struct snd_kcontrol_new cs4341_controls[] = {
+	SOC_DOUBLE_R_TLV("Master Playback Volume",
+			 CS4341_REG_VOLA, CS4341_REG_VOLB, 0, 90, 1, out_tlv),
+	SOC_ENUM("De-Emphasis Control", deemph_enum),
+	SOC_ENUM("Soft Ramp Zero Cross Control", srzc_enum),
+	SOC_SINGLE("Auto-Mute Switch", CS4341_REG_MODE2, 7, 1, 0),
+	SOC_SINGLE("Popguard Transient Switch", CS4341_REG_MODE2, 1, 1, 0),
+};
+
+static const struct snd_soc_dai_ops cs4341_dai_ops = {
+	.set_fmt	= cs4341_set_fmt,
+	.hw_params	= cs4341_hw_params,
+	.digital_mute	= cs4341_digital_mute,
+};
+
+static struct snd_soc_dai_driver cs4341_dai = {
+	.name			= "cs4341a-hifi",
+	.playback		= {
+		.stream_name	= "DAC Playback",
+		.channels_min	= 1,
+		.channels_max	= 2,
+		.rates		= SNDRV_PCM_RATE_8000_96000,
+		.formats	= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.ops			= &cs4341_dai_ops,
+	.symmetric_rates	= 1,
+};
+
+static const struct snd_soc_component_driver soc_component_cs4341 = {
+	.controls		= cs4341_controls,
+	.num_controls		= ARRAY_SIZE(cs4341_controls),
+	.dapm_widgets		= cs4341_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(cs4341_dapm_widgets),
+	.dapm_routes		= cs4341_routes,
+	.num_dapm_routes	= ARRAY_SIZE(cs4341_routes),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static const struct of_device_id __maybe_unused cs4341_dt_ids[] = {
+	{ .compatible = "cirrus,cs4341a", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, cs4341_dt_ids);
+
+static int cs4341_probe(struct device *dev)
+{
+	struct cs4341_priv *cs4341 = dev_get_drvdata(dev);
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(cs4341_reg_defaults); i++)
+		regmap_write(cs4341->regmap, cs4341_reg_defaults[i].reg,
+			     cs4341_reg_defaults[i].def);
+
+	return devm_snd_soc_register_component(dev, &soc_component_cs4341,
+					       &cs4341_dai, 1);
+}
+
+#if defined(CONFIG_I2C)
+static int cs4341_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct cs4341_priv *cs4341;
+
+	cs4341 = devm_kzalloc(&i2c->dev, sizeof(*cs4341), GFP_KERNEL);
+	if (!cs4341)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, cs4341);
+
+	cs4341->regcfg.reg_bits		= 8;
+	cs4341->regcfg.val_bits		= 8;
+	cs4341->regcfg.max_register	= CS4341_REG_VOLB;
+	cs4341->regcfg.cache_type	= REGCACHE_FLAT;
+	cs4341->regcfg.reg_defaults	= cs4341_reg_defaults;
+	cs4341->regcfg.num_reg_defaults	= ARRAY_SIZE(cs4341_reg_defaults);
+	cs4341->regmap = devm_regmap_init_i2c(i2c, &cs4341->regcfg);
+	if (IS_ERR(cs4341->regmap))
+		return PTR_ERR(cs4341->regmap);
+
+	return cs4341_probe(&i2c->dev);
+}
+
+static const struct i2c_device_id cs4341_i2c_id[] = {
+	{ "cs4341", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, cs4341_i2c_id);
+
+static struct i2c_driver cs4341_i2c_driver = {
+	.driver = {
+		.name = "cs4341-i2c",
+		.of_match_table = of_match_ptr(cs4341_dt_ids),
+	},
+	.probe = cs4341_i2c_probe,
+	.id_table = cs4341_i2c_id,
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static bool cs4341_reg_readable(struct device *dev, unsigned int reg)
+{
+	return false;
+}
+
+static int cs4341_spi_probe(struct spi_device *spi)
+{
+	struct cs4341_priv *cs4341;
+	int ret;
+
+	cs4341 = devm_kzalloc(&spi->dev, sizeof(*cs4341), GFP_KERNEL);
+	if (!cs4341)
+		return -ENOMEM;
+
+	if (!spi->bits_per_word)
+		spi->bits_per_word = 8;
+	if (!spi->max_speed_hz)
+		spi->max_speed_hz = 6000000;
+	ret = spi_setup(spi);
+	if (ret)
+		return ret;
+
+	spi_set_drvdata(spi, cs4341);
+
+	cs4341->regcfg.reg_bits		= 16;
+	cs4341->regcfg.val_bits		= 8;
+	cs4341->regcfg.write_flag_mask	= 0x20;
+	cs4341->regcfg.max_register	= CS4341_REG_VOLB;
+	cs4341->regcfg.cache_type	= REGCACHE_FLAT;
+	cs4341->regcfg.readable_reg	= cs4341_reg_readable;
+	cs4341->regcfg.reg_defaults	= cs4341_reg_defaults;
+	cs4341->regcfg.num_reg_defaults	= ARRAY_SIZE(cs4341_reg_defaults);
+	cs4341->regmap = devm_regmap_init_spi(spi, &cs4341->regcfg);
+	if (IS_ERR(cs4341->regmap))
+		return PTR_ERR(cs4341->regmap);
+
+	return cs4341_probe(&spi->dev);
+}
+
+static struct spi_driver cs4341_spi_driver = {
+	.driver = {
+		.name = "cs4341-spi",
+		.of_match_table = of_match_ptr(cs4341_dt_ids),
+	},
+	.probe = cs4341_spi_probe,
+};
+#endif
+
+static int __init cs4341_init(void)
+{
+	int ret = 0;
+
+#if defined(CONFIG_I2C)
+	ret = i2c_add_driver(&cs4341_i2c_driver);
+	if (ret)
+		return ret;
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&cs4341_spi_driver);
+#endif
+
+	return ret;
+}
+module_init(cs4341_init);
+
+static void __exit cs4341_exit(void)
+{
+#if defined(CONFIG_I2C)
+	i2c_del_driver(&cs4341_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&cs4341_spi_driver);
+#endif
+}
+module_exit(cs4341_exit);
+
+MODULE_AUTHOR("Alexander Shiyan <shc_work@mail.ru>");
+MODULE_DESCRIPTION("Cirrus Logic CS4341 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");

sound/soc/codecs/es8316.c

@@ -15,12 +15,14 @@
 #include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/mod_devicetable.h>
+#include <linux/mutex.h>
 #include <linux/regmap.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
+#include <sound/jack.h>
 #include "es8316.h"
 
 /* In slave mode at single speed, the codec is documented as accepting 5
@@ -33,6 +35,11 @@ static const unsigned int supported_mclk_lrck_ratios[] = {
 };
 
 struct es8316_priv {
+	struct mutex lock;
+	struct regmap *regmap;
+	struct snd_soc_component *component;
+	struct snd_soc_jack *jack;
+	int irq;
 	unsigned int sysclk;
 	unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS];
 	struct snd_pcm_hw_constraint_list sysclk_constraints;
@@ -94,6 +101,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0),
 	SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0),
 	SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0),
+	SOC_SINGLE("DAC Mono Mix Switch", ES8316_DAC_SET3, 3, 1, 0),
 
 	SOC_ENUM("Capture Polarity", adcpol),
 	SOC_SINGLE("Mic Boost Switch", ES8316_ADC_D2SEPGA, 0, 1, 0),
@@ -529,8 +537,162 @@ static struct snd_soc_dai_driver es8316_dai = {
 	.symmetric_rates = 1,
 };
 
+static void es8316_enable_micbias_for_mic_gnd_short_detect(
+	struct snd_soc_component *component)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+
+	snd_soc_dapm_mutex_lock(dapm);
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "Bias");
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "Analog power");
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "Mic Bias");
+	snd_soc_dapm_sync_unlocked(dapm);
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	msleep(20);
+}
+
+static void es8316_disable_micbias_for_mic_gnd_short_detect(
+	struct snd_soc_component *component)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+
+	snd_soc_dapm_mutex_lock(dapm);
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Bias");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Analog power");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Bias");
+	snd_soc_dapm_sync_unlocked(dapm);
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static irqreturn_t es8316_irq(int irq, void *data)
+{
+	struct es8316_priv *es8316 = data;
+	struct snd_soc_component *comp = es8316->component;
+	unsigned int flags;
+
+	mutex_lock(&es8316->lock);
+
+	regmap_read(es8316->regmap, ES8316_GPIO_FLAG, &flags);
+	if (flags == 0x00)
+		goto out; /* Powered-down / reset */
+
+	/* Catch spurious IRQ before set_jack is called */
+	if (!es8316->jack)
+		goto out;
+
+	dev_dbg(comp->dev, "gpio flags %#04x\n", flags);
+	if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) {
+		/* Jack removed, or spurious IRQ? */
+		if (es8316->jack->status & SND_JACK_MICROPHONE)
+			es8316_disable_micbias_for_mic_gnd_short_detect(comp);
+
+		if (es8316->jack->status & SND_JACK_HEADPHONE) {
+			snd_soc_jack_report(es8316->jack, 0,
+					    SND_JACK_HEADSET | SND_JACK_BTN_0);
+			dev_dbg(comp->dev, "jack unplugged\n");
+		}
+	} else if (!(es8316->jack->status & SND_JACK_HEADPHONE)) {
+		/* Jack inserted, determine type */
+		es8316_enable_micbias_for_mic_gnd_short_detect(comp);
+		regmap_read(es8316->regmap, ES8316_GPIO_FLAG, &flags);
+		dev_dbg(comp->dev, "gpio flags %#04x\n", flags);
+		if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) {
+			/* Jack unplugged underneath us */
+			es8316_disable_micbias_for_mic_gnd_short_detect(comp);
+		} else if (flags & ES8316_GPIO_FLAG_GM_NOT_SHORTED) {
+			/* Open, headset */
+			snd_soc_jack_report(es8316->jack,
+					    SND_JACK_HEADSET,
+					    SND_JACK_HEADSET);
+			/* Keep mic-gnd-short detection on for button press */
+		} else {
+			/* Shorted, headphones */
+			snd_soc_jack_report(es8316->jack,
+					    SND_JACK_HEADPHONE,
+					    SND_JACK_HEADSET);
+			/* No longer need mic-gnd-short detection */
+			es8316_disable_micbias_for_mic_gnd_short_detect(comp);
+		}
+	} else if (es8316->jack->status & SND_JACK_MICROPHONE) {
+		/* Interrupt while jack inserted, report button state */
+		if (flags & ES8316_GPIO_FLAG_GM_NOT_SHORTED) {
+			/* Open, button release */
+			snd_soc_jack_report(es8316->jack, 0, SND_JACK_BTN_0);
+		} else {
+			/* Short, button press */
+			snd_soc_jack_report(es8316->jack,
+					    SND_JACK_BTN_0,
+					    SND_JACK_BTN_0);
+		}
+	}
+
+out:
+	mutex_unlock(&es8316->lock);
+	return IRQ_HANDLED;
+}
+
+static void es8316_enable_jack_detect(struct snd_soc_component *component,
+				      struct snd_soc_jack *jack)
+{
+	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
+
+	mutex_lock(&es8316->lock);
+
+	es8316->jack = jack;
+
+	if (es8316->jack->status & SND_JACK_MICROPHONE)
+		es8316_enable_micbias_for_mic_gnd_short_detect(component);
+
+	snd_soc_component_update_bits(component, ES8316_GPIO_DEBOUNCE,
+				      ES8316_GPIO_ENABLE_INTERRUPT,
+				      ES8316_GPIO_ENABLE_INTERRUPT);
+
+	mutex_unlock(&es8316->lock);
+
+	/* Enable irq and sync initial jack state */
+	enable_irq(es8316->irq);
+	es8316_irq(es8316->irq, es8316);
+}
+
+static void es8316_disable_jack_detect(struct snd_soc_component *component)
+{
+	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
+
+	disable_irq(es8316->irq);
+
+	mutex_lock(&es8316->lock);
+
+	snd_soc_component_update_bits(component, ES8316_GPIO_DEBOUNCE,
+				      ES8316_GPIO_ENABLE_INTERRUPT, 0);
+
+	if (es8316->jack->status & SND_JACK_MICROPHONE) {
+		es8316_disable_micbias_for_mic_gnd_short_detect(component);
+		snd_soc_jack_report(es8316->jack, 0, SND_JACK_BTN_0);
+	}
+
+	es8316->jack = NULL;
+
+	mutex_unlock(&es8316->lock);
+}
+
+static int es8316_set_jack(struct snd_soc_component *component,
+			   struct snd_soc_jack *jack, void *data)
+{
+	if (jack)
+		es8316_enable_jack_detect(component, jack);
+	else
+		es8316_disable_jack_detect(component);
+
+	return 0;
+}
+
 static int es8316_probe(struct snd_soc_component *component)
 {
+	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
+
+	es8316->component = component;
+
 	/* Reset codec and enable current state machine */
 	snd_soc_component_write(component, ES8316_RESET, 0x3f);
 	usleep_range(5000, 5500);
@@ -555,6 +717,7 @@ static int es8316_probe(struct snd_soc_component *component)
 
 static const struct snd_soc_component_driver soc_component_dev_es8316 = {
 	.probe			= es8316_probe,
+	.set_jack		= es8316_set_jack,
 	.controls		= es8316_snd_controls,
 	.num_controls		= ARRAY_SIZE(es8316_snd_controls),
 	.dapm_widgets		= es8316_dapm_widgets,
@@ -566,18 +729,29 @@ static const struct snd_soc_component_driver soc_component_dev_es8316 = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static const struct regmap_range es8316_volatile_ranges[] = {
+	regmap_reg_range(ES8316_GPIO_FLAG, ES8316_GPIO_FLAG),
+};
+
+static const struct regmap_access_table es8316_volatile_table = {
+	.yes_ranges	= es8316_volatile_ranges,
+	.n_yes_ranges	= ARRAY_SIZE(es8316_volatile_ranges),
+};
+
 static const struct regmap_config es8316_regmap = {
 	.reg_bits = 8,
 	.val_bits = 8,
 	.max_register = 0x53,
+	.volatile_table	= &es8316_volatile_table,
 	.cache_type = REGCACHE_RBTREE,
 };
 
 static int es8316_i2c_probe(struct i2c_client *i2c_client,
 			    const struct i2c_device_id *id)
 {
+	struct device *dev = &i2c_client->dev;
 	struct es8316_priv *es8316;
-	struct regmap *regmap;
+	int ret;
 
 	es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv),
 			      GFP_KERNEL);
@@ -586,9 +760,23 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client,
 
 	i2c_set_clientdata(i2c_client, es8316);
 
-	regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap);
-	if (IS_ERR(regmap))
-		return PTR_ERR(regmap);
+	es8316->regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap);
+	if (IS_ERR(es8316->regmap))
+		return PTR_ERR(es8316->regmap);
+
+	es8316->irq = i2c_client->irq;
+	mutex_init(&es8316->lock);
+
+	ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq,
+					IRQF_TRIGGER_HIGH | IRQF_ONESHOT,
+					"es8316", es8316);
+	if (ret == 0) {
+		/* Gets re-enabled by es8316_set_jack() */
+		disable_irq(es8316->irq);
+	} else {
+		dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret);
+		es8316->irq = -ENXIO;
+	}
 
 	return devm_snd_soc_register_component(&i2c_client->dev,
 				      &soc_component_dev_es8316,

sound/soc/codecs/es8316.h

@@ -126,4 +126,11 @@
 #define ES8316_SERDATA2_LEN_16		0x0c
 #define ES8316_SERDATA2_LEN_32		0x10
 
+/* ES8316_GPIO_DEBOUNCE	*/
+#define ES8316_GPIO_ENABLE_INTERRUPT		0x02
+
+/* ES8316_GPIO_FLAG */
+#define ES8316_GPIO_FLAG_GM_NOT_SHORTED		0x02
+#define ES8316_GPIO_FLAG_HP_NOT_INSERTED	0x04
+
 #endif

sound/soc/codecs/pcm512x.c

@@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
 		if (ret != 0) {
 			dev_err(component->dev,
 				"Failed to set digital mute: %d\n", ret);
-			mutex_unlock(&pcm512x->mutex);
-			return ret;
+			goto unlock;
 		}
 
 		regmap_read_poll_timeout(pcm512x->regmap,
 					 PCM512x_ANALOG_MUTE_DET,
 					 mute_det, (mute_det & 0x3) == 0,
 					 200, 10000);
-
-		mutex_unlock(&pcm512x->mutex);
 	} else {
 		pcm512x->mute &= ~0x1;
 		ret = pcm512x_update_mute(pcm512x);
 		if (ret != 0) {
 			dev_err(component->dev,
 				"Failed to update digital mute: %d\n", ret);
-			mutex_unlock(&pcm512x->mutex);
-			return ret;
+			goto unlock;
 		}
 
 		regmap_read_poll_timeout(pcm512x->regmap,
@@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
 					 200, 10000);
 	}
 
+unlock:
 	mutex_unlock(&pcm512x->mutex);
 
-	return 0;
+	return ret;
 }
 
 static const struct snd_soc_dai_ops pcm512x_dai_ops = {

sound/soc/codecs/rk3328_codec.h

+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * rk3328 ALSA SoC Audio driver
+ *
+ * Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved.
+ */
+
+#ifndef _RK3328_CODEC_H
+#define _RK3328_CODEC_H
+
+#include <linux/bitfield.h>
+
+/* codec register */
+#define CODEC_RESET			(0x00 << 2)
+#define DAC_INIT_CTRL1			(0x03 << 2)
+#define DAC_INIT_CTRL2			(0x04 << 2)
+#define DAC_INIT_CTRL3			(0x05 << 2)
+#define DAC_PRECHARGE_CTRL		(0x22 << 2)
+#define DAC_PWR_CTRL			(0x23 << 2)
+#define DAC_CLK_CTRL			(0x24 << 2)
+#define HPMIX_CTRL			(0x25 << 2)
+#define DAC_SELECT			(0x26 << 2)
+#define HPOUT_CTRL			(0x27 << 2)
+#define HPOUTL_GAIN_CTRL		(0x28 << 2)
+#define HPOUTR_GAIN_CTRL		(0x29 << 2)
+#define HPOUT_POP_CTRL			(0x2a << 2)
+
+/* REG00: CODEC_RESET */
+#define PWR_RST_BYPASS_DIS		(0x0 << 6)
+#define PWR_RST_BYPASS_EN		(0x1 << 6)
+#define DIG_CORE_RST			(0x0 << 1)
+#define DIG_CORE_WORK			(0x1 << 1)
+#define SYS_RST				(0x0 << 0)
+#define SYS_WORK			(0x1 << 0)
+
+/* REG03: DAC_INIT_CTRL1 */
+#define PIN_DIRECTION_MASK		BIT(5)
+#define PIN_DIRECTION_IN		(0x0 << 5)
+#define PIN_DIRECTION_OUT		(0x1 << 5)
+#define DAC_I2S_MODE_MASK		BIT(4)
+#define DAC_I2S_MODE_SLAVE		(0x0 << 4)
+#define DAC_I2S_MODE_MASTER		(0x1 << 4)
+
+/* REG04: DAC_INIT_CTRL2 */
+#define DAC_I2S_LRP_MASK		BIT(7)
+#define DAC_I2S_LRP_NORMAL		(0x0 << 7)
+#define DAC_I2S_LRP_REVERSAL		(0x1 << 7)
+#define DAC_VDL_MASK			GENMASK(6, 5)
+#define DAC_VDL_16BITS			(0x0 << 5)
+#define DAC_VDL_20BITS			(0x1 << 5)
+#define DAC_VDL_24BITS			(0x2 << 5)
+#define DAC_VDL_32BITS			(0x3 << 5)
+#define DAC_MODE_MASK			GENMASK(4, 3)
+#define DAC_MODE_RJM			(0x0 << 3)
+#define DAC_MODE_LJM			(0x1 << 3)
+#define DAC_MODE_I2S			(0x2 << 3)
+#define DAC_MODE_PCM			(0x3 << 3)
+#define DAC_LR_SWAP_MASK		BIT(2)
+#define DAC_LR_SWAP_DIS			(0x0 << 2)
+#define DAC_LR_SWAP_EN			(0x1 << 2)
+
+/* REG05: DAC_INIT_CTRL3 */
+#define DAC_WL_MASK			GENMASK(3, 2)
+#define DAC_WL_16BITS			(0x0 << 2)
+#define DAC_WL_20BITS			(0x1 << 2)
+#define DAC_WL_24BITS			(0x2 << 2)
+#define DAC_WL_32BITS			(0x3 << 2)
+#define DAC_RST_MASK			BIT(1)
+#define DAC_RST_EN			(0x0 << 1)
+#define DAC_RST_DIS			(0x1 << 1)
+#define DAC_BCP_MASK			BIT(0)
+#define DAC_BCP_NORMAL			(0x0 << 0)
+#define DAC_BCP_REVERSAL		(0x1 << 0)
+
+/* REG22: DAC_PRECHARGE_CTRL */
+#define DAC_CHARGE_XCHARGE_MASK		BIT(7)
+#define DAC_CHARGE_DISCHARGE		(0x0 << 7)
+#define DAC_CHARGE_PRECHARGE		(0x1 << 7)
+#define DAC_CHARGE_CURRENT_64I_MASK	BIT(6)
+#define DAC_CHARGE_CURRENT_64I		(0x1 << 6)
+#define DAC_CHARGE_CURRENT_32I_MASK	BIT(5)
+#define DAC_CHARGE_CURRENT_32I		(0x1 << 5)
+#define DAC_CHARGE_CURRENT_16I_MASK	BIT(4)
+#define DAC_CHARGE_CURRENT_16I		(0x1 << 4)
+#define DAC_CHARGE_CURRENT_08I_MASK	BIT(3)
+#define DAC_CHARGE_CURRENT_08I		(0x1 << 3)
+#define DAC_CHARGE_CURRENT_04I_MASK	BIT(2)
+#define DAC_CHARGE_CURRENT_04I		(0x1 << 2)
+#define DAC_CHARGE_CURRENT_02I_MASK	BIT(1)
+#define DAC_CHARGE_CURRENT_02I		(0x1 << 1)
+#define DAC_CHARGE_CURRENT_I_MASK	BIT(0)
+#define DAC_CHARGE_CURRENT_I		(0x1 << 0)
+#define DAC_CHARGE_CURRENT_ALL_MASK	GENMASK(6, 0)
+#define DAC_CHARGE_CURRENT_ALL_OFF	0x00
+#define DAC_CHARGE_CURRENT_ALL_ON	0x7f
+
+/* REG23: DAC_PWR_CTRL */
+#define DAC_PWR_MASK			BIT(6)
+#define DAC_PWR_OFF			(0x0 << 6)
+#define DAC_PWR_ON			(0x1 << 6)
+#define DACL_PATH_REFV_MASK		BIT(5)
+#define DACL_PATH_REFV_OFF		(0x0 << 5)
+#define DACL_PATH_REFV_ON		(0x1 << 5)
+#define HPOUTL_ZERO_CROSSING_MASK	BIT(4)
+#define HPOUTL_ZERO_CROSSING_OFF	(0x0 << 4)
+#define HPOUTL_ZERO_CROSSING_ON		(0x1 << 4)
+#define DACR_PATH_REFV_MASK		BIT(1)
+#define DACR_PATH_REFV_OFF		(0x0 << 1)
+#define DACR_PATH_REFV_ON		(0x1 << 1)
+#define HPOUTR_ZERO_CROSSING_MASK	BIT(0)
+#define HPOUTR_ZERO_CROSSING_OFF	(0x0 << 0)
+#define HPOUTR_ZERO_CROSSING_ON		(0x1 << 0)
+
+/* REG24: DAC_CLK_CTRL */
+#define DACL_REFV_MASK			BIT(7)
+#define DACL_REFV_OFF			(0x0 << 7)
+#define DACL_REFV_ON			(0x1 << 7)
+#define DACL_CLK_MASK			BIT(6)
+#define DACL_CLK_OFF			(0x0 << 6)
+#define DACL_CLK_ON			(0x1 << 6)
+#define DACL_MASK			BIT(5)
+#define DACL_OFF			(0x0 << 5)
+#define DACL_ON				(0x1 << 5)
+#define DACL_INIT_MASK			BIT(4)
+#define DACL_INIT_OFF			(0x0 << 4)
+#define DACL_INIT_ON			(0x1 << 4)
+#define DACR_REFV_MASK			BIT(3)
+#define DACR_REFV_OFF			(0x0 << 3)
+#define DACR_REFV_ON			(0x1 << 3)
+#define DACR_CLK_MASK			BIT(2)
+#define DACR_CLK_OFF			(0x0 << 2)
+#define DACR_CLK_ON			(0x1 << 2)
+#define DACR_MASK			BIT(1)
+#define DACR_OFF			(0x0 << 1)
+#define DACR_ON				(0x1 << 1)
+#define DACR_INIT_MASK			BIT(0)
+#define DACR_INIT_OFF			(0x0 << 0)
+#define DACR_INIT_ON			(0x1 << 0)
+
+/* REG25: HPMIX_CTRL*/
+#define HPMIXL_MASK			BIT(6)
+#define HPMIXL_DIS			(0x0 << 6)
+#define HPMIXL_EN			(0x1 << 6)
+#define HPMIXL_INIT_MASK		BIT(5)
+#define HPMIXL_INIT_DIS			(0x0 << 5)
+#define HPMIXL_INIT_EN			(0x1 << 5)
+#define HPMIXL_INIT2_MASK		BIT(4)
+#define HPMIXL_INIT2_DIS		(0x0 << 4)
+#define HPMIXL_INIT2_EN			(0x1 << 4)
+#define HPMIXR_MASK			BIT(2)
+#define HPMIXR_DIS			(0x0 << 2)
+#define HPMIXR_EN			(0x1 << 2)
+#define HPMIXR_INIT_MASK		BIT(1)
+#define HPMIXR_INIT_DIS			(0x0 << 1)
+#define HPMIXR_INIT_EN			(0x1 << 1)
+#define HPMIXR_INIT2_MASK		BIT(0)
+#define HPMIXR_INIT2_DIS		(0x0 << 0)
+#define HPMIXR_INIT2_EN			(0x1 << 0)
+
+/* REG26: DAC_SELECT */
+#define DACL_SELECT_MASK		BIT(4)
+#define DACL_UNSELECT			(0x0 << 4)
+#define DACL_SELECT			(0x1 << 4)
+#define DACR_SELECT_MASK		BIT(0)
+#define DACR_UNSELECT			(0x0 << 0)
+#define DACR_SELECT			(0x1 << 0)
+
+/* REG27: HPOUT_CTRL */
+#define HPOUTL_MASK			BIT(7)
+#define HPOUTL_DIS			(0x0 << 7)
+#define HPOUTL_EN			(0x1 << 7)
+#define HPOUTL_INIT_MASK		BIT(6)
+#define HPOUTL_INIT_DIS			(0x0 << 6)
+#define HPOUTL_INIT_EN			(0x1 << 6)
+#define HPOUTL_MUTE_MASK		BIT(5)
+#define HPOUTL_MUTE			(0x0 << 5)
+#define HPOUTL_UNMUTE			(0x1 << 5)
+#define HPOUTR_MASK			BIT(4)
+#define HPOUTR_DIS			(0x0 << 4)
+#define HPOUTR_EN			(0x1 << 4)
+#define HPOUTR_INIT_MASK		BIT(3)
+#define HPOUTR_INIT_DIS			(0x0 << 3)
+#define HPOUTR_INIT_EN			(0x1 << 3)
+#define HPOUTR_MUTE_MASK		BIT(2)
+#define HPOUTR_MUTE			(0x0 << 2)
+#define HPOUTR_UNMUTE			(0x1 << 2)
+
+/* REG28: HPOUTL_GAIN_CTRL */
+#define HPOUTL_GAIN_MASK		GENMASK(4, 0)
+
+/* REG29: HPOUTR_GAIN_CTRL */
+#define HPOUTR_GAIN_MASK		GENMASK(4, 0)
+
+/* REG2a: HPOUT_POP_CTRL */
+#define HPOUTR_POP_MASK			GENMASK(5, 4)
+#define HPOUTR_POP_XCHARGE		(0x1 << 4)
+#define HPOUTR_POP_WORK			(0x2 << 4)
+#define HPOUTL_POP_MASK			GENMASK(1, 0)
+#define HPOUTL_POP_XCHARGE		(0x1 << 0)
+#define HPOUTL_POP_WORK			(0x2 << 0)
+
+#define RK3328_HIFI			0
+
+struct rk3328_reg_msk_val {
+	unsigned int reg;
+	unsigned int msk;
+	unsigned int val;
+};
+
+#endif

sound/soc/codecs/rt274.c

@@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
-	regmap_read(rt274->regmap,
+	ret = regmap_read(rt274->regmap,
 		RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+	if (ret)
+		return ret;
+
 	if (val != RT274_VENDOR_ID) {
 		dev_err(&i2c->dev,
 			"Device with ID register %#x is not rt274\n", val);

sound/soc/codecs/rt5682.c

@@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 	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);
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
 
 	mutex_unlock(&rt5682->calibrate_mutex);
 

sound/soc/codecs/wm8904.c

@@ -1837,9 +1837,6 @@ static int wm8904_set_bias_level(struct snd_soc_component *component,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		ret = clk_prepare_enable(wm8904->mclk);
-		if (ret)
-			return ret;
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
@@ -1864,6 +1861,15 @@ static int wm8904_set_bias_level(struct snd_soc_component *component,
 				return ret;
 			}
 
+			ret = clk_prepare_enable(wm8904->mclk);
+			if (ret) {
+				dev_err(component->dev,
+					"Failed to enable MCLK: %d\n", ret);
+				regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
+						       wm8904->supplies);
+				return ret;
+			}
+
 			regcache_cache_only(wm8904->regmap, false);
 			regcache_sync(wm8904->regmap);
 
@@ -2108,16 +2114,13 @@ static const struct regmap_config wm8904_regmap = {
 };
 
 #ifdef CONFIG_OF
-static enum wm8904_type wm8904_data = WM8904;
-static enum wm8904_type wm8912_data = WM8912;
-
 static const struct of_device_id wm8904_of_match[] = {
 	{
 		.compatible = "wlf,wm8904",
-		.data = &wm8904_data,
+		.data = (void *)WM8904,
 	}, {
 		.compatible = "wlf,wm8912",
-		.data = &wm8912_data,
+		.data = (void *)WM8912,
 	}, {
 		/* sentinel */
 	}
@@ -2158,7 +2161,7 @@ static int wm8904_i2c_probe(struct i2c_client *i2c,
 		match = of_match_node(wm8904_of_match, i2c->dev.of_node);
 		if (match == NULL)
 			return -EINVAL;
-		wm8904->devtype = *((enum wm8904_type *)match->data);
+		wm8904->devtype = (enum wm8904_type)match->data;
 	} else {
 		wm8904->devtype = id->driver_data;
 	}

sound/soc/generic/Kconfig

@@ -8,14 +8,6 @@ config SND_SIMPLE_CARD
 	  This option enables generic simple sound card support
 	  It also support DPCM of multi CPU single Codec ststem.
 
-config SND_SIMPLE_SCU_CARD
-	tristate "ASoC Simple SCU sound card support"
-	depends on OF
-	select SND_SIMPLE_CARD_UTILS
-	help
-	  This option enables generic simple SCU sound card support.
-	  It supports DPCM of multi CPU single Codec system.
-
 config SND_AUDIO_GRAPH_CARD
 	tristate "ASoC Audio Graph sound card support"
 	depends on OF
@@ -24,12 +16,3 @@ config SND_AUDIO_GRAPH_CARD
 	  This option enables generic simple sound card support
 	  with OF-graph DT bindings.
 	  It also support DPCM of multi CPU single Codec ststem.
-
-config SND_AUDIO_GRAPH_SCU_CARD
-	tristate "ASoC Audio Graph SCU sound card support"
-	depends on OF
-	select SND_SIMPLE_CARD_UTILS
-	help
-	  This option enables generic simple SCU sound card support
-	  with OF-graph DT bindings.
-	  It supports DPCM of multi CPU single Codec ststem.

sound/soc/generic/Makefile

 # SPDX-License-Identifier: GPL-2.0
 snd-soc-simple-card-utils-objs	:= simple-card-utils.o
 snd-soc-simple-card-objs	:= simple-card.o
-snd-soc-simple-scu-card-objs	:= simple-scu-card.o
 snd-soc-audio-graph-card-objs	:= audio-graph-card.o
-snd-soc-audio-graph-scu-card-objs	:= audio-graph-scu-card.o
 
 obj-$(CONFIG_SND_SIMPLE_CARD_UTILS)	+= snd-soc-simple-card-utils.o
 obj-$(CONFIG_SND_SIMPLE_CARD)		+= snd-soc-simple-card.o
-obj-$(CONFIG_SND_SIMPLE_SCU_CARD)	+= snd-soc-simple-scu-card.o
 obj-$(CONFIG_SND_AUDIO_GRAPH_CARD)	+= snd-soc-audio-graph-card.o
-obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD)	+= snd-soc-audio-graph-scu-card.o

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

@@ -283,12 +283,20 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep)
 	/* use endpoint/port reg if exist */
 	ret = of_graph_parse_endpoint(ep, &info);
 	if (ret == 0) {
-		if (info.id)
+		/*
+		 * Because it will count port/endpoint if it doesn't have "reg".
+		 * But, we can't judge whether it has "no reg", or "reg = <0>"
+		 * only of_graph_parse_endpoint().
+		 * We need to check "reg" property
+		 */
+		if (of_get_property(ep,   "reg", NULL))
 			return info.id;
-		if (info.port)
+
+		node = of_get_parent(ep);
+		of_node_put(node);
+		if (of_get_property(node, "reg", NULL))
 			return info.port;
 	}
-
 	node = of_graph_get_port_parent(ep);
 
 	/*

sound/soc/intel/Kconfig

@@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI
 config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
 	default ACPI
-	depends on X86 && ACPI
+	depends on X86 && ACPI && PCI
 	select SND_SST_IPC_ACPI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH

sound/soc/intel/atom/sst-atom-controls.c

@@ -647,7 +647,7 @@ static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w,
 		set_mixer = false;
 	}
 
-	if (set_mixer == false)
+	if (!set_mixer)
 		return 0;
 
 	if (SND_SOC_DAPM_EVENT_ON(event) ||

sound/soc/intel/atom/sst-mfld-platform-pcm.c

@@ -190,7 +190,7 @@ int sst_fill_stream_params(void *substream,
 	map = ctx->pdata->pdev_strm_map;
 	map_size = ctx->pdata->strm_map_size;
 
-	if (is_compress == true)
+	if (is_compress)
 		cstream = (struct snd_compr_stream *)substream;
 	else
 		pstream = (struct snd_pcm_substream *)substream;

sound/soc/intel/atom/sst/sst_acpi.c

@@ -255,18 +255,16 @@ static int is_byt(void)
 	return status;
 }
 
-static int is_byt_cr(struct device *dev, bool *bytcr)
+static bool is_byt_cr(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
 	int status = 0;
 
-	if (IS_ENABLED(CONFIG_IOSF_MBI)) {
-		u32 bios_status;
-
-		if (!is_byt() || !iosf_mbi_available()) {
-			/* bail silently */
-			return status;
-		}
+	if (!is_byt())
+		return false;
 
+	if (iosf_mbi_available()) {
+		u32 bios_status;
 		status = iosf_mbi_read(BT_MBI_UNIT_PMC, /* 0x04 PUNIT */
 				       MBI_REG_READ, /* 0x10 */
 				       0x006, /* BIOS_CONFIG */
@@ -278,15 +276,28 @@ static int is_byt_cr(struct device *dev, bool *bytcr)
 			/* bits 26:27 mirror PMIC options */
 			bios_status = (bios_status >> 26) & 3;
 
-			if ((bios_status == 1) || (bios_status == 3))
-				*bytcr = true;
-			else
-				dev_info(dev, "BYT-CR not detected\n");
+			if (bios_status == 1 || bios_status == 3) {
+				dev_info(dev, "Detected Baytrail-CR platform\n");
+				return true;
+			}
+
+			dev_info(dev, "BYT-CR not detected\n");
 		}
 	} else {
-		dev_info(dev, "IOSF_MBI not enabled, no BYT-CR detection\n");
+		dev_info(dev, "IOSF_MBI not available, no BYT-CR detection\n");
 	}
-	return status;
+
+	if (platform_get_resource(pdev, IORESOURCE_IRQ, 5) == NULL) {
+		/*
+		 * Some devices detected as BYT-T have only a single IRQ listed,
+		 * causing platform_get_irq with index 5 to return -ENXIO.
+		 * The correct IRQ in this case is at index 0, as on BYT-CR.
+		 */
+		dev_info(dev, "Falling back to Baytrail-CR platform\n");
+		return true;
+	}
+
+	return false;
 }
 
 
@@ -301,7 +312,6 @@ static int sst_acpi_probe(struct platform_device *pdev)
 	struct platform_device *plat_dev;
 	struct sst_platform_info *pdata;
 	unsigned int dev_id;
-	bool bytcr = false;
 
 	id = acpi_match_device(dev->driver->acpi_match_table, dev);
 	if (!id)
@@ -333,10 +343,7 @@ static int sst_acpi_probe(struct platform_device *pdev)
 	if (ret < 0)
 		return ret;
 
-	ret = is_byt_cr(dev, &bytcr);
-	if (!((ret < 0) || (bytcr == false))) {
-		dev_info(dev, "Detected Baytrail-CR platform\n");
-
+	if (is_byt_cr(pdev)) {
 		/* override resource info */
 		byt_rvp_platform_data.res_info = &bytcr_res_info;
 	}

sound/soc/intel/atom/sst/sst_drv_interface.c

@@ -146,7 +146,7 @@ static int sst_power_control(struct device *dev, bool state)
 	int ret = 0;
 	int usage_count = 0;
 
-	if (state == true) {
+	if (state) {
 		ret = pm_runtime_get_sync(dev);
 		usage_count = GET_USAGE_COUNT(dev);
 		dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count);

sound/soc/intel/atom/sst/sst_loader.c

@@ -269,7 +269,7 @@ static void sst_do_memcpy(struct list_head *memcpy_list)
 	struct sst_memcpy_list *listnode;
 
 	list_for_each_entry(listnode, memcpy_list, memcpylist) {
-		if (listnode->is_io == true)
+		if (listnode->is_io)
 			memcpy32_toio((void __iomem *)listnode->dstn,
 					listnode->src, listnode->size);
 		else

sound/soc/intel/baytrail/sst-baytrail-ipc.c

@@ -278,7 +278,6 @@ static int sst_byt_process_notification(struct sst_byt *byt,
 	struct sst_byt_stream *stream;
 	u64 header;
 	u8 msg_id, stream_id;
-	int handled = 1;
 
 	header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
 	msg_id = sst_byt_header_msg_id(header);
@@ -298,7 +297,7 @@ static int sst_byt_process_notification(struct sst_byt *byt,
 		break;
 	}
 
-	return handled;
+	return 1;
 }
 
 static irqreturn_t sst_byt_irq_thread(int irq, void *context)

sound/soc/intel/baytrail/sst-baytrail-pcm.c

@@ -188,7 +188,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		sst_byt_stream_start(byt, pcm_data->stream, 0);
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
-		if (pdata->restore_stream == true)
+		if (pdata->restore_stream)
 			schedule_work(&pcm_data->work);
 		else
 			sst_byt_stream_resume(byt, pcm_data->stream);

sound/soc/intel/boards/broadwell.c

@@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},

sound/soc/intel/boards/bytcr_rt5640.c

@@ -428,6 +428,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"),
+		},
+		.driver_data = (void *)(BYT_RT5640_IN1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),

sound/soc/intel/boards/glk_rt5682_max98357a.c

@@ -164,7 +164,7 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }

sound/soc/intel/boards/haswell.c

@@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},

sound/soc/intel/boards/kbl_da7219_max98927.c

@@ -221,7 +221,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		rate->min = rate->max = 48000;
 		channels->min = channels->max = 2;
 		snd_mask_none(fmt);
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	}
 
 	/*
@@ -229,7 +229,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * thus changing the mask here
 	 */
 	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
 }

sound/soc/intel/common/soc-acpi-intel-byt-match.c

@@ -154,6 +154,15 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.sof_tplg_filename = "intel/sof-byt-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
+	{
+		.id = "ESSX8316",
+		.drv_name = "bytcht_es8316",
+		.fw_filename = "intel/fw_sst_0f28.bin",
+		.board = "bytcht_es8316",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-es8316.tplg",
+		.asoc_plat_name = "sst-mfld-platform",
+	},
 	/* some Baytrail platforms rely on RT5645, use CHT machine driver */
 	{
 		.id = "10EC5645",

sound/soc/intel/haswell/sst-haswell-ipc.c

@@ -1216,7 +1216,7 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 		return ret;
 	}
 
-	stream->commited = 1;
+	stream->commited = true;
 	trace_hsw_stream_alloc_reply(stream);
 
 	return 0;

sound/soc/intel/haswell/sst-haswell-pcm.c

@@ -544,7 +544,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
 		dev_err(rtd->dev, "error: invalid DAI ID %d\n",
 			rtd->cpu_dai->id);
 		return -EINVAL;
-	};
+	}
 
 	ret = sst_hsw_stream_format(hsw, pcm_data->stream,
 		path_id, stream_type, SST_HSW_STREAM_FORMAT_PCM_FORMAT);
@@ -861,7 +861,7 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream)
 		dev_dbg(rtd->dev, "error: free stream failed %d\n", ret);
 		goto out;
 	}
-	pcm_data->allocated = 0;
+	pcm_data->allocated = false;
 	pcm_data->stream = NULL;
 
 out:

sound/soc/intel/skylake/skl-messages.c

@@ -416,7 +416,7 @@ int skl_resume_dsp(struct skl *skl)
 	snd_hdac_ext_bus_ppcap_int_enable(bus, true);
 
 	/* check if DSP 1st boot is done */
-	if (skl->skl_sst->is_first_boot == true)
+	if (skl->skl_sst->is_first_boot)
 		return 0;
 
 	/*

sound/soc/intel/skylake/skl-pcm.c

@@ -1423,7 +1423,7 @@ static int skl_platform_soc_probe(struct snd_soc_component *component)
 		if (!ops)
 			return -EIO;
 
-		if (skl->skl_sst->is_first_boot == false) {
+		if (!skl->skl_sst->is_first_boot) {
 			dev_err(component->dev, "DSP reports first boot done!!!\n");
 			return -EIO;
 		}

sound/soc/intel/skylake/skl-topology.c

@@ -3103,7 +3103,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be,
 	ac->size = dfw_ac->max;
 
 	if (ac->max) {
-		ac->params = (char *) devm_kzalloc(dev, ac->max, GFP_KERNEL);
+		ac->params = devm_kzalloc(dev, ac->max, GFP_KERNEL);
 		if (!ac->params)
 			return -ENOMEM;
 

sound/soc/sh/dma-sh7760.c

@@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
 
-	pr_debug("PCM data: addr 0x%08ulx len %d\n",
+	pr_debug("PCM data: addr 0x%08lx len %d\n",
 		 (u32)runtime->dma_addr, runtime->dma_bytes);
  
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {