Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/imx',...

Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/imx', 'asoc/topic/intel', 'asoc/topic/jz4740' and 'asoc/topic/max98357a' into asoc-next

Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt

+Ingenic JZ4740 I2S controller
+
+Required properties:
+- compatible : "ingenic,jz4740-i2s"
+- reg : I2S registers location and length
+- clocks : AIC and I2S PLL clock specifiers.
+- clock-names: "aic" and "i2s"
+- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels
+- dma-names: Must be "tx" and "rx"
+
+Example:
+
+i2s: i2s@10020000 {
+	compatible = "ingenic,jz4740-i2s";
+	reg = <0x10020000 0x94>;
+
+	clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>;
+	clock-names = "aic", "i2s";
+
+	dmas = <&dma 2>, <&dma 3>;
+	dma-names = "tx", "rx";
+
+};

Documentation/devicetree/bindings/sound/max98357a.txt

+Maxim MAX98357A audio DAC
+
+This node models the Maxim MAX98357A DAC.
+
+Required properties:
+- compatible   : "maxim,max98357a"
+- sdmode-gpios : GPIO specifier for the GPIO -> DAC SDMODE pin
+
+Example:
+
+max98357a {
+	compatible = "maxim,max98357a";
+	sdmode-gpios = <&qcom_pinmux 25 0>;
+};

sound/soc/codecs/Kconfig

@@ -69,6 +69,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX98090 if I2C
 	select SND_SOC_MAX98095 if I2C
+	select SND_SOC_MAX98357A
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
@@ -456,6 +457,9 @@ config SND_SOC_MAX98090
 config SND_SOC_MAX98095
        tristate
 
+config SND_SOC_MAX98357A
+       tristate
+
 config SND_SOC_MAX9850
 	tristate
 

sound/soc/codecs/Makefile

@@ -64,6 +64,7 @@ snd-soc-max9768-objs := max9768.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-max98090-objs := max98090.o
 snd-soc-max98095-objs := max98095.o
+snd-soc-max98357a-objs := max98357a.o
 snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
@@ -245,6 +246,7 @@ obj-$(CONFIG_SND_SOC_MAX9768)	+= snd-soc-max9768.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98090)	+= snd-soc-max98090.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
+obj-$(CONFIG_SND_SOC_MAX98357A)	+= snd-soc-max98357a.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o

sound/soc/codecs/max98357a.c

+/* Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ *
+ * max98357a.c -- MAX98357A ALSA SoC Codec driver
+ */
+
+#include <linux/module.h>
+#include <linux/gpio.h>
+#include <sound/soc.h>
+
+#define DRV_NAME "max98357a"
+
+static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_dai *dai)
+{
+	struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		gpiod_set_value(sdmode, 1);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		gpiod_set_value(sdmode, 0);
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("SDMode", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUTPUT("Speaker"),
+};
+
+static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
+	{"Speaker", NULL, "SDMode"},
+};
+
+static int max98357a_codec_probe(struct snd_soc_codec *codec)
+{
+	struct gpio_desc *sdmode;
+
+	sdmode = devm_gpiod_get(codec->dev, "sdmode");
+	if (IS_ERR(sdmode)) {
+		dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n",
+				__func__, PTR_ERR(sdmode));
+		return PTR_ERR(sdmode);
+	}
+	gpiod_direction_output(sdmode, 0);
+	snd_soc_codec_set_drvdata(codec, sdmode);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver max98357a_codec_driver = {
+	.probe			= max98357a_codec_probe,
+	.dapm_widgets		= max98357a_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(max98357a_dapm_widgets),
+	.dapm_routes		= max98357a_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(max98357a_dapm_routes),
+};
+
+static struct snd_soc_dai_ops max98357a_dai_ops = {
+	.trigger	= max98357a_daiops_trigger,
+};
+
+static struct snd_soc_dai_driver max98357a_dai_driver = {
+	.name = DRV_NAME,
+	.playback = {
+		.stream_name	= DRV_NAME "-playback",
+		.formats	= SNDRV_PCM_FMTBIT_S16 |
+					SNDRV_PCM_FMTBIT_S24 |
+					SNDRV_PCM_FMTBIT_S32,
+		.rates		= SNDRV_PCM_RATE_8000 |
+					SNDRV_PCM_RATE_16000 |
+					SNDRV_PCM_RATE_48000 |
+					SNDRV_PCM_RATE_96000,
+		.rate_min	= 8000,
+		.rate_max	= 96000,
+		.channels_min	= 1,
+		.channels_max	= 2,
+	},
+	.ops    = &max98357a_dai_ops,
+};
+
+static int max98357a_platform_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = snd_soc_register_codec(&pdev->dev, &max98357a_codec_driver,
+			&max98357a_dai_driver, 1);
+	if (ret)
+		dev_err(&pdev->dev, "%s() error registering codec driver: %d\n",
+				__func__, ret);
+
+	return ret;
+}
+
+static int max98357a_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id max98357a_device_id[] = {
+	{ .compatible = "maxim," DRV_NAME, },
+	{}
+};
+MODULE_DEVICE_TABLE(of, max98357a_device_id);
+#endif
+
+static struct platform_driver max98357a_platform_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.of_match_table = of_match_ptr(max98357a_device_id),
+	},
+	.probe	= max98357a_platform_probe,
+	.remove = max98357a_platform_remove,
+};
+module_platform_driver(max98357a_platform_driver);
+
+MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);

sound/soc/codecs/rt5645.h

@@ -47,6 +47,7 @@
 #define RT5645_STO_DAC_MIXER			0x2a
 #define RT5645_MONO_DAC_MIXER			0x2b
 #define RT5645_DIG_MIXER			0x2c
+#define RT5650_A_DAC_SOUR			0x2d
 #define RT5645_DIG_INF1_DATA			0x2f
 /* Mixer - PDM */
 #define RT5645_PDM_OUT_CTRL			0x31
@@ -150,6 +151,8 @@
 #define RT5645_IL_CMD				0xdb
 #define RT5645_IL_CMD2				0xdc
 #define RT5645_IL_CMD3				0xdd
+#define RT5650_4BTN_IL_CMD1			0xdf
+#define RT5650_4BTN_IL_CMD2			0xe0
 #define RT5645_DRC1_HL_CTRL1			0xe7
 #define RT5645_DRC2_HL_CTRL1			0xe9
 #define RT5645_MUTI_DRC_CTRL1			0xea
@@ -472,6 +475,12 @@
 #define RT5645_DAC_L2_DAC_R_VOL_MASK		(0x1 << 4)
 #define RT5645_DAC_L2_DAC_R_VOL_SFT		4
 
+/* Analog DAC1/2 Input Source Control (0x2d) */
+#define RT5650_A_DAC1_L_IN_SFT			3
+#define RT5650_A_DAC1_R_IN_SFT			2
+#define RT5650_A_DAC2_L_IN_SFT			1
+#define RT5650_A_DAC2_R_IN_SFT			0
+
 /* Digital Interface Data Control (0x2f) */
 #define RT5645_IF1_ADC2_IN_SEL			(0x1 << 15)
 #define RT5645_IF1_ADC2_IN_SFT			15
@@ -1111,50 +1120,27 @@
 #define RT5645_DMIC_2_M_NOR			(0x0 << 8)
 #define RT5645_DMIC_2_M_ASYN			(0x1 << 8)
 
+/* ASRC clock source selection (0x84, 0x85) */
+#define RT5645_CLK_SEL_SYS			(0x0)
+#define RT5645_CLK_SEL_I2S1_ASRC		(0x1)
+#define RT5645_CLK_SEL_I2S2_ASRC		(0x2)
+#define RT5645_CLK_SEL_SYS2			(0x5)
+
 /* ASRC Control 2 (0x84) */
-#define RT5645_MDA_L_M_MASK			(0x1 << 15)
-#define RT5645_MDA_L_M_SFT			15
-#define RT5645_MDA_L_M_NOR			(0x0 << 15)
-#define RT5645_MDA_L_M_ASYN			(0x1 << 15)
-#define RT5645_MDA_R_M_MASK			(0x1 << 14)
-#define RT5645_MDA_R_M_SFT			14
-#define RT5645_MDA_R_M_NOR			(0x0 << 14)
-#define RT5645_MDA_R_M_ASYN			(0x1 << 14)
-#define RT5645_MAD_L_M_MASK			(0x1 << 13)
-#define RT5645_MAD_L_M_SFT			13
-#define RT5645_MAD_L_M_NOR			(0x0 << 13)
-#define RT5645_MAD_L_M_ASYN			(0x1 << 13)
-#define RT5645_MAD_R_M_MASK			(0x1 << 12)
-#define RT5645_MAD_R_M_SFT			12
-#define RT5645_MAD_R_M_NOR			(0x0 << 12)
-#define RT5645_MAD_R_M_ASYN			(0x1 << 12)
-#define RT5645_ADC_M_MASK			(0x1 << 11)
-#define RT5645_ADC_M_SFT			11
-#define RT5645_ADC_M_NOR			(0x0 << 11)
-#define RT5645_ADC_M_ASYN			(0x1 << 11)
-#define RT5645_STO_DAC_M_MASK			(0x1 << 5)
-#define RT5645_STO_DAC_M_SFT			5
-#define RT5645_STO_DAC_M_NOR			(0x0 << 5)
-#define RT5645_STO_DAC_M_ASYN			(0x1 << 5)
-#define RT5645_I2S1_R_D_MASK			(0x1 << 4)
-#define RT5645_I2S1_R_D_SFT			4
-#define RT5645_I2S1_R_D_DIS			(0x0 << 4)
-#define RT5645_I2S1_R_D_EN			(0x1 << 4)
-#define RT5645_I2S2_R_D_MASK			(0x1 << 3)
-#define RT5645_I2S2_R_D_SFT			3
-#define RT5645_I2S2_R_D_DIS			(0x0 << 3)
-#define RT5645_I2S2_R_D_EN			(0x1 << 3)
-#define RT5645_PRE_SCLK_MASK			(0x3)
-#define RT5645_PRE_SCLK_SFT			0
-#define RT5645_PRE_SCLK_512			(0x0)
-#define RT5645_PRE_SCLK_1024			(0x1)
-#define RT5645_PRE_SCLK_2048			(0x2)
+#define RT5645_DA_STO_CLK_SEL_MASK		(0xf << 12)
+#define RT5645_DA_STO_CLK_SEL_SFT		12
+#define RT5645_DA_MONOL_CLK_SEL_MASK		(0xf << 8)
+#define RT5645_DA_MONOL_CLK_SEL_SFT		8
+#define RT5645_DA_MONOR_CLK_SEL_MASK		(0xf << 4)
+#define RT5645_DA_MONOR_CLK_SEL_SFT		4
+#define RT5645_AD_STO1_CLK_SEL_MASK		(0xf << 0)
+#define RT5645_AD_STO1_CLK_SEL_SFT		0
 
 /* ASRC Control 3 (0x85) */
-#define RT5645_I2S1_RATE_MASK			(0xf << 12)
-#define RT5645_I2S1_RATE_SFT			12
-#define RT5645_I2S2_RATE_MASK			(0xf << 8)
-#define RT5645_I2S2_RATE_SFT			8
+#define RT5645_AD_MONOL_CLK_SEL_MASK		(0xf << 4)
+#define RT5645_AD_MONOL_CLK_SEL_SFT		4
+#define RT5645_AD_MONOR_CLK_SEL_MASK		(0xf << 0)
+#define RT5645_AD_MONOR_CLK_SEL_SFT		0
 
 /* ASRC Control 4 (0x89) */
 #define RT5645_I2S1_PD_MASK			(0x7 << 12)
@@ -2175,6 +2161,24 @@ enum {
 	RT5645_DMIC_DATA_GPIO11,
 };
 
+enum {
+	CODEC_TYPE_RT5645,
+	CODEC_TYPE_RT5650,
+};
+
+/* filter mask */
+enum {
+	RT5645_DA_STEREO_FILTER = 0x1,
+	RT5645_DA_MONO_L_FILTER = (0x1 << 1),
+	RT5645_DA_MONO_R_FILTER = (0x1 << 2),
+	RT5645_AD_STEREO_FILTER = (0x1 << 3),
+	RT5645_AD_MONO_L_FILTER = (0x1 << 4),
+	RT5645_AD_MONO_R_FILTER = (0x1 << 5),
+};
+
+int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec,
+		unsigned int filter_mask, unsigned int clk_src);
+
 struct rt5645_priv {
 	struct snd_soc_codec *codec;
 	struct rt5645_platform_data pdata;
@@ -2184,6 +2188,7 @@ struct rt5645_priv {
 	struct snd_soc_jack *mic_jack;
 	struct delayed_work jack_detect_work;
 
+	int codec_type;
 	int sysclk;
 	int sysclk_src;
 	int lrck[RT5645_AIFS];

sound/soc/codecs/rt5670.c

@@ -592,6 +592,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source,
 	return 0;
 }
 
+
+/**
+ * rt5670_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @codec: SoC audio codec device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5670 can
+ * only support standard 32fs or 64fs i2s format, ASRC should be enabled to
+ * support special i2s clock format such as Intel's 100fs(100 * sampling rate).
+ * ASRC function will track i2s clock and generate a corresponding system clock
+ * for codec. This function provides an API to select the clock source for a
+ * set of filters specified by the mask. And the codec driver will turn on ASRC
+ * for these filters if ASRC is selected as their clock source.
+ */
+int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec,
+			    unsigned int filter_mask, unsigned int clk_src)
+{
+	unsigned int asrc2_mask = 0, asrc2_value = 0;
+	unsigned int asrc3_mask = 0, asrc3_value = 0;
+
+	if (clk_src > RT5670_CLK_SEL_SYS3)
+		return -EINVAL;
+
+	if (filter_mask & RT5670_DA_STEREO_FILTER) {
+		asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_DA_STO_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_DA_MONO_L_FILTER) {
+		asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_DA_MONOL_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_DA_MONO_R_FILTER) {
+		asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_DA_MONOR_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_AD_STEREO_FILTER) {
+		asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_AD_STO1_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_AD_MONO_L_FILTER) {
+		asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK;
+		asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_AD_MONOL_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_AD_MONO_R_FILTER)  {
+		asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK;
+		asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_AD_MONOR_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_UP_RATE_FILTER) {
+		asrc3_mask |= RT5670_UP_CLK_SEL_MASK;
+		asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_UP_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5670_DOWN_RATE_FILTER) {
+		asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK;
+		asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK)
+				| (clk_src <<  RT5670_DOWN_CLK_SEL_SFT);
+	}
+
+	if (asrc2_mask)
+		snd_soc_update_bits(codec, RT5670_ASRC_2,
+				    asrc2_mask, asrc2_value);
+
+	if (asrc3_mask)
+		snd_soc_update_bits(codec, RT5670_ASRC_3,
+				    asrc3_mask, asrc3_value);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src);
+
 /* Digital Mixer */
 static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = {
 	SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER,

sound/soc/codecs/rt5670.h

@@ -1023,50 +1023,33 @@
 #define RT5670_DMIC_2_M_NOR			(0x0 << 8)
 #define RT5670_DMIC_2_M_ASYN			(0x1 << 8)
 
+/* ASRC clock source selection (0x84, 0x85) */
+#define RT5670_CLK_SEL_SYS			(0x0)
+#define RT5670_CLK_SEL_I2S1_ASRC		(0x1)
+#define RT5670_CLK_SEL_I2S2_ASRC		(0x2)
+#define RT5670_CLK_SEL_I2S3_ASRC		(0x3)
+#define RT5670_CLK_SEL_SYS2			(0x5)
+#define RT5670_CLK_SEL_SYS3			(0x6)
+
 /* ASRC Control 2 (0x84) */
-#define RT5670_MDA_L_M_MASK			(0x1 << 15)
-#define RT5670_MDA_L_M_SFT			15
-#define RT5670_MDA_L_M_NOR			(0x0 << 15)
-#define RT5670_MDA_L_M_ASYN			(0x1 << 15)
-#define RT5670_MDA_R_M_MASK			(0x1 << 14)
-#define RT5670_MDA_R_M_SFT			14
-#define RT5670_MDA_R_M_NOR			(0x0 << 14)
-#define RT5670_MDA_R_M_ASYN			(0x1 << 14)
-#define RT5670_MAD_L_M_MASK			(0x1 << 13)
-#define RT5670_MAD_L_M_SFT			13
-#define RT5670_MAD_L_M_NOR			(0x0 << 13)
-#define RT5670_MAD_L_M_ASYN			(0x1 << 13)
-#define RT5670_MAD_R_M_MASK			(0x1 << 12)
-#define RT5670_MAD_R_M_SFT			12
-#define RT5670_MAD_R_M_NOR			(0x0 << 12)
-#define RT5670_MAD_R_M_ASYN			(0x1 << 12)
-#define RT5670_ADC_M_MASK			(0x1 << 11)
-#define RT5670_ADC_M_SFT			11
-#define RT5670_ADC_M_NOR			(0x0 << 11)
-#define RT5670_ADC_M_ASYN			(0x1 << 11)
-#define RT5670_STO_DAC_M_MASK			(0x1 << 5)
-#define RT5670_STO_DAC_M_SFT			5
-#define RT5670_STO_DAC_M_NOR			(0x0 << 5)
-#define RT5670_STO_DAC_M_ASYN			(0x1 << 5)
-#define RT5670_I2S1_R_D_MASK			(0x1 << 4)
-#define RT5670_I2S1_R_D_SFT			4
-#define RT5670_I2S1_R_D_DIS			(0x0 << 4)
-#define RT5670_I2S1_R_D_EN			(0x1 << 4)
-#define RT5670_I2S2_R_D_MASK			(0x1 << 3)
-#define RT5670_I2S2_R_D_SFT			3
-#define RT5670_I2S2_R_D_DIS			(0x0 << 3)
-#define RT5670_I2S2_R_D_EN			(0x1 << 3)
-#define RT5670_PRE_SCLK_MASK			(0x3)
-#define RT5670_PRE_SCLK_SFT			0
-#define RT5670_PRE_SCLK_512			(0x0)
-#define RT5670_PRE_SCLK_1024			(0x1)
-#define RT5670_PRE_SCLK_2048			(0x2)
+#define RT5670_DA_STO_CLK_SEL_MASK		(0xf << 12)
+#define RT5670_DA_STO_CLK_SEL_SFT		12
+#define RT5670_DA_MONOL_CLK_SEL_MASK		(0xf << 8)
+#define RT5670_DA_MONOL_CLK_SEL_SFT		8
+#define RT5670_DA_MONOR_CLK_SEL_MASK		(0xf << 4)
+#define RT5670_DA_MONOR_CLK_SEL_SFT		4
+#define RT5670_AD_STO1_CLK_SEL_MASK		(0xf << 0)
+#define RT5670_AD_STO1_CLK_SEL_SFT		0
 
 /* ASRC Control 3 (0x85) */
-#define RT5670_I2S1_RATE_MASK			(0xf << 12)
-#define RT5670_I2S1_RATE_SFT			12
-#define RT5670_I2S2_RATE_MASK			(0xf << 8)
-#define RT5670_I2S2_RATE_SFT			8
+#define RT5670_UP_CLK_SEL_MASK			(0xf << 12)
+#define RT5670_UP_CLK_SEL_SFT			12
+#define RT5670_DOWN_CLK_SEL_MASK		(0xf << 8)
+#define RT5670_DOWN_CLK_SEL_SFT			8
+#define RT5670_AD_MONOL_CLK_SEL_MASK		(0xf << 4)
+#define RT5670_AD_MONOL_CLK_SEL_SFT		4
+#define RT5670_AD_MONOR_CLK_SEL_MASK		(0xf << 0)
+#define RT5670_AD_MONOR_CLK_SEL_SFT		0
 
 /* ASRC Control 4 (0x89) */
 #define RT5670_I2S1_PD_MASK			(0x7 << 12)
@@ -1983,6 +1966,21 @@ enum {
 	RT5670_DMIC_DATA_GPIO5,
 };
 
+/* filter mask */
+enum {
+	RT5670_DA_STEREO_FILTER = 0x1,
+	RT5670_DA_MONO_L_FILTER = (0x1 << 1),
+	RT5670_DA_MONO_R_FILTER = (0x1 << 2),
+	RT5670_AD_STEREO_FILTER = (0x1 << 3),
+	RT5670_AD_MONO_L_FILTER = (0x1 << 4),
+	RT5670_AD_MONO_R_FILTER = (0x1 << 5),
+	RT5670_UP_RATE_FILTER   = (0x1 << 6),
+	RT5670_DOWN_RATE_FILTER = (0x1 << 7),
+};
+
+int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec,
+			    unsigned int filter_mask, unsigned int clk_src);
+
 struct rt5670_priv {
 	struct snd_soc_codec *codec;
 	struct rt5670_platform_data pdata;

sound/soc/fsl/fsl_spdif.c

@@ -90,7 +90,6 @@ struct spdif_mixer_control {
  * @sysclk: system clock for rx clock rate measurement
  * @dma_params_tx: DMA parameters for transmit channel
  * @dma_params_rx: DMA parameters for receive channel
- * @name: driver name
  */
 struct fsl_spdif_priv {
 	struct spdif_mixer_control fsl_spdif_control;
@@ -109,12 +108,8 @@ struct fsl_spdif_priv {
 	struct clk *sysclk;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
-
-	/* The name space will be allocated dynamically */
-	char name[0];
 };
 
-
 /* DPLL locked and lock loss interrupt handler */
 static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
 {
@@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	if (!np)
 		return -ENODEV;
 
-	spdif_priv = devm_kzalloc(&pdev->dev,
-			sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1,
-			GFP_KERNEL);
+	spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL);
 	if (!spdif_priv)
 		return -ENOMEM;
 
-	strcpy(spdif_priv->name, np->name);
-
 	spdif_priv->pdev = pdev;
 
 	/* Initialize this copy of the CPU DAI driver structure */
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
-	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+	spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev);
 
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -1203,7 +1194,7 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	}
 
 	ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
-			spdif_priv->name, spdif_priv);
+			       dev_name(&pdev->dev), spdif_priv);
 	if (ret) {
 		dev_err(&pdev->dev, "could not claim irq %u\n", irq);
 		return ret;

sound/soc/fsl/fsl_ssi.c

@@ -160,7 +160,7 @@ struct fsl_ssi_soc_data {
  */
 struct fsl_ssi_private {
 	struct regmap *regs;
-	unsigned int irq;
+	int irq;
 	struct snd_soc_dai_driver cpu_dai_drv;
 
 	unsigned int dai_fmt;

sound/soc/fsl/imx-spdif.c

@@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
 	data->card.dev = &pdev->dev;
 	data->card.dai_link = &data->dai;
 	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
 
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret)

sound/soc/intel/Kconfig

@@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
           platforms with RT5672 audio codec.
           Say Y if you have such a device
           If unsure select "N".
+
+config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
+	tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec"
+	depends on X86_INTEL_LPSS
+	select SND_SOC_RT5645
+	select SND_SST_MFLD_PLATFORM
+	select SND_SST_IPC_ACPI
+	help
+	  This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
+	  platforms with RT5645 audio codec.
+	  If unsure select "N".

sound/soc/intel/Makefile

@@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
 snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
+snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
 
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
@@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
 
 # DSP driver
 obj-$(CONFIG_SND_SST_IPC) += sst/

sound/soc/intel/broadwell.c

@@ -140,8 +140,6 @@ static struct snd_soc_ops broadwell_rt286_ops = {
 
 static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
 	struct sst_hsw *broadwell = pdata->dsp;
 	int ret;
@@ -155,14 +153,6 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
 		return ret;
 	}
 
-	/* always connected - check HP for jack detect */
-	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
-	snd_soc_dapm_enable_pin(dapm, "Speaker");
-	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
-	snd_soc_dapm_enable_pin(dapm, "Line Jack");
-	snd_soc_dapm_enable_pin(dapm, "DMIC1");
-	snd_soc_dapm_enable_pin(dapm, "DMIC2");
-
 	return 0;
 }
 

sound/soc/intel/byt-rt5640.c

@@ -132,7 +132,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
 	struct snd_soc_codec *codec = runtime->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_card *card = runtime->card;
 	const struct snd_soc_dapm_route *custom_map;
 	int num_routes;
@@ -161,7 +160,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 		num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map);
 	}
 
-	ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes);
+	ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes);
 	if (ret)
 		return ret;
 
@@ -171,13 +170,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 			return ret;
 	}
 
-	snd_soc_dapm_ignore_suspend(dapm, "HPOL");
-	snd_soc_dapm_ignore_suspend(dapm, "HPOR");
-
-	snd_soc_dapm_ignore_suspend(dapm, "SPOLP");
-	snd_soc_dapm_ignore_suspend(dapm, "SPOLN");
-	snd_soc_dapm_ignore_suspend(dapm, "SPORP");
-	snd_soc_dapm_ignore_suspend(dapm, "SPORN");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
 
 	return ret;
 }

sound/soc/intel/cht_bsw_rt5645.c

+/*
+ *  cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
+ *                     Cherrytrail and Braswell, with RT5645 codec.
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  Author: Fang, Yang A <yang.a.fang@intel.com>
+ *	        N,Harshapriya <harshapriya.n@intel.com>
+ *  This file is modified from cht_bsw_rt5672.c
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5645.h"
+#include "sst-atom-controls.h"
+
+#define CHT_PLAT_CLK_3_HZ	19200000
+#define CHT_CODEC_DAI	"rt5645-aif1"
+
+struct cht_mc_private {
+	struct snd_soc_jack hp_jack;
+	struct snd_soc_jack mic_jack;
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+	int i;
+
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_pcm_runtime *rtd;
+
+		rtd = card->rtd + i;
+		if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+			     strlen(CHT_CODEC_DAI)))
+			return rtd->codec_dai;
+	}
+	return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret;
+
+	codec_dai = cht_get_codec_dai(card);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+		return -EIO;
+	}
+
+	if (!SND_SOC_DAPM_EVENT_OFF(event))
+		return 0;
+
+	/* Set codec sysclk source to its internal clock because codec PLL will
+	 * be off when idle and MCLK will also be off by ACPI when codec is
+	 * runtime suspended. Codec needs clock for jack detection and button
+	 * press.
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
+			0, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"DMIC L1", NULL, "Int Mic"},
+	{"DMIC R1", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOL"},
+	{"Ext Spk", NULL, "SPOR"},
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx" },
+	{"codec_in1", NULL, "ssp2 Rx" },
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+	{"Headphone", NULL, "Platform Clock"},
+	{"Headset Mic", NULL, "Platform Clock"},
+	{"Int Mic", NULL, "Platform Clock"},
+	{"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+				  CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
+				params_rate(params) * 512, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
+
+	/* Select clk_i2s1_asrc as ASRC clock source */
+	rt5645_sel_asrc_clk_src(codec,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_DA_MONO_L_FILTER |
+				RT5645_DA_MONO_R_FILTER |
+				RT5645_AD_STEREO_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+
+	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+	if (ret < 0) {
+		dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_jack_new(codec, "Headphone Jack",
+				SND_JACK_HEADPHONE,
+				&ctx->hp_jack);
+	if (ret) {
+		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_jack_new(codec, "Mic Jack",
+				SND_JACK_MICROPHONE,
+				&ctx->mic_jack);
+	if (ret) {
+		dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);
+
+	return ret;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+			    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);
+
+	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+	.startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+	.hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5645-aif1",
+		.codec_name = "i2c-10EC5645:00",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+					| SND_SOC_DAIFMT_CBS_CFS,
+		.init = cht_codec_init,
+		.be_hw_params_fixup = cht_codec_fixup,
+		.ignore_suspend = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+	.name = "chtrt5645",
+	.dai_link = cht_dailink,
+	.num_links = ARRAY_SIZE(cht_dailink),
+	.dapm_widgets = cht_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+	.dapm_routes = cht_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	struct cht_mc_private *drv;
+
+	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
+	if (!drv)
+		return -ENOMEM;
+
+	snd_soc_card_cht.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_cht);
+	return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+	.driver = {
+		.name = "cht-bsw-rt5645",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
+MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5645");

sound/soc/intel/cht_bsw_rt5672.c

@@ -140,6 +140,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
 	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
 
 	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
 	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
@@ -148,6 +149,19 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
 		return ret;
 	}
 
+	/* Select codec ASRC clock source to track I2S1 clock, because codec
+	 * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
+	 * be supported by RT5672. Otherwise, ASRC will be disabled and cause
+	 * noise.
+	 */
+	rt5670_sel_asrc_clk_src(codec,
+				RT5670_DA_STEREO_FILTER
+				| RT5670_DA_MONO_L_FILTER
+				| RT5670_DA_MONO_R_FILTER
+				| RT5670_AD_STEREO_FILTER
+				| RT5670_AD_MONO_L_FILTER
+				| RT5670_AD_MONO_R_FILTER,
+				RT5670_CLK_SEL_I2S1_ASRC);
 	return 0;
 }
 

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

@@ -320,11 +320,6 @@ static struct snd_pcm_ops sst_byt_pcm_ops = {
 	.mmap		= sst_byt_pcm_mmap,
 };
 
-static void sst_byt_pcm_free(struct snd_pcm *pcm)
-{
-	snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
 static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_pcm *pcm = rtd->pcm;
@@ -403,7 +398,6 @@ static struct snd_soc_platform_driver byt_soc_platform = {
 	.remove		= sst_byt_pcm_remove,
 	.ops		= &sst_byt_pcm_ops,
 	.pcm_new	= sst_byt_pcm_new,
-	.pcm_free	= sst_byt_pcm_free,
 };
 
 static const struct snd_soc_component_driver byt_dai_component = {

sound/soc/intel/sst-dsp.c

@@ -410,7 +410,6 @@ void sst_dsp_free(struct sst_dsp *sst)
 	if (sst->ops->free)
 		sst->ops->free(sst);
 
-	if (sst->dma)
 	sst_dma_free(sst->dma);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_free);

sound/soc/intel/sst-firmware.c

@@ -791,6 +791,7 @@ int sst_module_alloc_blocks(struct sst_module *module)
 	struct sst_block_allocator ba;
 	int ret;
 
+	memset(&ba, 0, sizeof(ba));
 	ba.size = module->size;
 	ba.type = module->type;
 	ba.offset = module->offset;
@@ -864,6 +865,7 @@ int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime,
 	if (module->persistent_size == 0)
 		return 0;
 
+	memset(&ba, 0, sizeof(ba));
 	ba.size = module->persistent_size;
 	ba.type = SST_MEM_DRAM;
 

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

@@ -306,7 +306,7 @@ static void hsw_reset(struct sst_dsp *sst)
 static int hsw_set_dsp_D0(struct sst_dsp *sst)
 {
 	int tries = 10;
-	u32 reg;
+	u32 reg, fw_dump_bit;
 
 	/* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
@@ -368,7 +368,9 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst)
 	can't be accessed, please enable each block before accessing. */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
 	reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
+	/* for D0, always enable the block(DSRAM[0]) used for FW dump */
+	fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT;
+	writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0);
 
 
 	/* disable DMA finish function for SSP0 & SSP1 */
@@ -491,6 +493,7 @@ static const struct sst_sram_shift sram_shift[] = {
 	{SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
 	{SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
 };
+
 static u32 hsw_block_get_bit(struct sst_mem_block *block)
 {
 	u32 bit = 0, shift = 0, index;
@@ -587,6 +590,8 @@ static int hsw_block_disable(struct sst_mem_block *block)
 
 	val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
 	bit = hsw_block_get_bit(block);
+	/* don't disable DSRAM[0], keep it always enable for FW dump*/
+	if (bit != (1 << SST_VDRTCL0_DSRAMPGE_SHIFT))
 		writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0);
 
 	/* wait 18 DSP clock ticks */
@@ -612,7 +617,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 	const struct sst_adsp_memregion *region;
 	struct device *dev;
 	int ret = -ENODEV, i, j, region_count;
-	u32 offset, size;
+	u32 offset, size, fw_dump_bit;
 
 	dev = sst->dma_dev;
 
@@ -669,9 +674,11 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 		}
 	}
 
+	/* always enable the block(DSRAM[0]) used for FW dump */
+	fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT;
 	/* set default power gating control, enable power gating control for all blocks. that is,
 	can't be accessed, please enable each block before accessing. */
-	writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
+	writel(0xffffffff & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0);
 
 	return 0;
 }

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

@@ -94,6 +94,8 @@
 /* Mailbox */
 #define IPC_MAX_MAILBOX_BYTES	256
 
+#define INVALID_STREAM_HW_ID	0xffffffff
+
 /* Global Message - Types and Replies */
 enum ipc_glb_type {
 	IPC_GLB_GET_FW_VERSION = 0,		/* Retrieves firmware version */
@@ -275,7 +277,6 @@ struct sst_hsw {
 	/* FW config */
 	struct sst_hsw_ipc_fw_ready fw_ready;
 	struct sst_hsw_ipc_fw_version version;
-	struct sst_module *scratch;
 	bool fw_done;
 	struct sst_fw *sst_fw;
 
@@ -337,12 +338,6 @@ static inline u32 msg_get_stage_type(u32 msg)
 	return (msg & IPC_STG_TYPE_MASK) >>  IPC_STG_TYPE_SHIFT;
 }
 
-static inline u32 msg_set_stage_type(u32 msg, u32 type)
-{
-	return (msg & ~IPC_STG_TYPE_MASK) +
-		(type << IPC_STG_TYPE_SHIFT);
-}
-
 static inline u32 msg_get_stream_id(u32 msg)
 {
 	return (msg & IPC_STR_ID_MASK) >>  IPC_STR_ID_SHIFT;
@@ -969,45 +964,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw,
 }
 
 /* Mixer Controls */
-int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel)
-{
-	int ret;
-
-	ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel,
-		&stream->mute_volume[channel]);
-	if (ret < 0)
-		return ret;
-
-	ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
-			stream->reply.stream_hw_id, channel);
-		return ret;
-	}
-
-	stream->mute[channel] = 1;
-	return 0;
-}
-
-int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel)
-
-{
-	int ret;
-
-	stream->mute[channel] = 0;
-	ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel,
-		stream->mute_volume[channel]);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
-			stream->reply.stream_hw_id, channel);
-		return ret;
-	}
-
-	return 0;
-}
-
 int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
 	u32 stage_id, u32 channel, u32 *volume)
 {
@@ -1021,17 +977,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream
 	return 0;
 }
 
-int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u64 curve_duration,
-	enum sst_hsw_volume_curve curve)
-{
-	/* curve duration in steps of 100ns */
-	stream->vol_req.curve_duration = curve_duration;
-	stream->vol_req.curve_type = curve;
-
-	return 0;
-}
-
 /* stream volume */
 int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume)
@@ -1083,42 +1028,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	return 0;
 }
 
-int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
-{
-	int ret;
-
-	ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel,
-		&hsw->mute_volume[channel]);
-	if (ret < 0)
-		return ret;
-
-	ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
-			channel);
-		return ret;
-	}
-
-	hsw->mute[channel] = 1;
-	return 0;
-}
-
-int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
-{
-	int ret;
-
-	ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel,
-		hsw->mixer_info.volume_register_address[channel]);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
-			channel);
-		return ret;
-	}
-
-	hsw->mute[channel] = 0;
-	return 0;
-}
-
 int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 *volume)
 {
@@ -1132,16 +1041,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	return 0;
 }
 
-int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
-	 u64 curve_duration, enum sst_hsw_volume_curve curve)
-{
-	/* curve duration in steps of 100ns */
-	hsw->curve_duration = curve_duration;
-	hsw->curve_type = curve;
-
-	return 0;
-}
-
 /* global mixer volume */
 int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 volume)
@@ -1208,6 +1107,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
 		return NULL;
 
 	spin_lock_irqsave(&sst->spinlock, flags);
+	stream->reply.stream_hw_id = INVALID_STREAM_HW_ID;
 	list_add(&stream->node, &hsw->stream_list);
 	stream->notify_position = notify_position;
 	stream->pdata = data;
@@ -1449,48 +1349,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 
 /* Stream Information - these calls could be inline but we want the IPC
  ABI to be opaque to client PCM drivers to cope with any future ABI changes */
-int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.stream_hw_id;
-}
-
-int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.mixer_hw_id;
-}
-
-u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.read_position_register_address;
-}
-
-u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.presentation_position_register_address;
-}
-
-u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel)
-{
-	if (channel >= 2)
-		return 0;
-
-	return stream->reply.peak_meter_register_address[channel];
-}
-
-u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel)
-{
-	if (channel >= 2)
-		return 0;
-
-	return stream->reply.volume_register_address[channel];
-}
-
 int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
 {
 	struct sst_hsw_ipc_stream_info_reply *reply;
@@ -1628,30 +1486,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
 	return ppos;
 }
 
-int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 stage_id, u32 position)
-{
-	u32 header;
-	int ret;
-
-	trace_stream_write_position(stream->reply.stream_hw_id, position);
-
-	header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
-		IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
-	header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
-	header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT);
-	header |= (stage_id << IPC_STG_ID_SHIFT);
-	stream->wpos.position = position;
-
-	ret = ipc_tx_message_nowait(hsw, header, &stream->wpos,
-		sizeof(stream->wpos));
-	if (ret < 0)
-		dev_err(hsw->dev, "error: stream %d set position %d failed\n",
-			stream->reply.stream_hw_id, position);
-
-	return ret;
-}
-
 /* physical BE config */
 int sst_hsw_device_set_config(struct sst_hsw *hsw,
 	enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
@@ -2132,7 +1966,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata)
 	dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE,
 			hsw->dx_context, hsw->dx_context_paddr);
 	sst_dsp_free(hsw->dsp);
-	kfree(hsw->scratch);
 	kthread_stop(hsw->tx_thread);
 	kfree(hsw->msg);
 }

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

@@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw,
 u32 create_channel_map(enum sst_hsw_channel_config config);
 
 /* Stream Mixer Controls - */
-int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel);
-int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel);
-
 int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume);
 int sst_hsw_stream_get_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume);
 
-int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u64 curve_duration,
-	enum sst_hsw_volume_curve curve);
-
 /* Global Mixer Controls - */
-int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
-int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
-
 int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 volume);
 int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 *volume);
 
-int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
-	u64 curve_duration, enum sst_hsw_volume_curve curve);
-
 /* Stream API */
 struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
 	u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data),
@@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 offset, u32 size);
 int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 offset, u32 size);
-int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel);
-u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel);
 int sst_hsw_mixer_get_info(struct sst_hsw *hsw);
 
 /* Stream ALSA trigger operations */
@@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 *position);
 int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 *position);
-int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 stage_id, u32 position);
 u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream);
 u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
@@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
 /* DX Config */
 int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx);
-int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
-	u32 *offset, u32 *size, u32 *source);
 
 /* init */
 int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata);

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

@@ -643,12 +643,6 @@ static struct snd_pcm_ops sst_platform_ops = {
 	.pointer = sst_platform_pcm_pointer,
 };
 
-static void sst_pcm_free(struct snd_pcm *pcm)
-{
-	dev_dbg(pcm->dev, "sst_pcm_free called\n");
-	snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
 static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *dai = rtd->cpu_dai;
@@ -679,7 +673,6 @@ static struct snd_soc_platform_driver sst_soc_platform_drv  = {
 	.ops		= &sst_platform_ops,
 	.compr_ops	= &sst_platform_compr_ops,
 	.pcm_new	= sst_pcm_new,
-	.pcm_free	= sst_pcm_free,
 };
 
 static const struct snd_soc_component_driver sst_component = {

sound/soc/intel/sst/sst_acpi.c

@@ -352,6 +352,8 @@ static struct sst_machines sst_acpi_bytcr[] = {
 static struct sst_machines sst_acpi_chv[] = {
 	{"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin",
 						&chv_platform_data },
+	{"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin",
+						&chv_platform_data },
 	{},
 };
 

sound/soc/intel/sst/sst_loader.c

@@ -324,7 +324,6 @@ void sst_firmware_load_cb(const struct firmware *fw, void *context)
 
 	if (ctx->sst_state != SST_RESET ||
 			ctx->fw_in_mem != NULL) {
-		if (fw != NULL)
 		release_firmware(fw);
 		mutex_unlock(&ctx->sst_lock);
 		return;

sound/soc/jz4740/jz4740-i2s.c

@@ -14,6 +14,8 @@
 
 #include <linux/init.h>
 #include <linux/io.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 #include <linux/kernel.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
@@ -83,6 +85,8 @@
 #define JZ_AIC_I2S_STATUS_BUSY BIT(2)
 
 #define JZ_AIC_CLK_DIV_MASK 0xf
+#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
 
 struct jz4740_i2s {
 	struct resource *mem;
@@ -237,10 +241,14 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	unsigned int sample_size;
-	uint32_t ctrl;
+	uint32_t ctrl, div_reg;
+	int div;
 
 	ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
 
+	div_reg = jz4740_i2s_read(i2s, JZ_REG_AIC_CLK_DIV);
+	div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
+
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		sample_size = 0;
@@ -264,7 +272,10 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 		ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
 	}
 
+	div_reg &= ~I2SDIV_DV_MASK;
+	div_reg |= (div - 1) << I2SDIV_DV_SHIFT;
 	jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
+	jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div_reg);
 
 	return 0;
 }
@@ -415,6 +426,13 @@ static const struct snd_soc_component_driver jz4740_i2s_component = {
 	.name		= "jz4740-i2s",
 };
 
+#ifdef CONFIG_OF
+static const struct of_device_id jz4740_of_matches[] = {
+	{ .compatible = "ingenic,jz4740-i2s" },
+	{ /* sentinel */ }
+};
+#endif
+
 static int jz4740_i2s_dev_probe(struct platform_device *pdev)
 {
 	struct jz4740_i2s *i2s;
@@ -455,6 +473,7 @@ static struct platform_driver jz4740_i2s_driver = {
 	.probe = jz4740_i2s_dev_probe,
 	.driver = {
 		.name = "jz4740-i2s",
+		.of_match_table = of_match_ptr(jz4740_of_matches)
 	},
 };