Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next

Documentation/devicetree/bindings/regmap/regmap.txt

+Device-Tree binding for regmap
+
+The endianness mode of CPU & Device scenarios:
+Index     Device     Endianness properties
+---------------------------------------------------
+1         BE         'big-endian'
+2         LE         'little-endian'
+
+For one device driver, which will run in different scenarios above
+on different SoCs using the devicetree, we need one way to simplify
+this.
+
+Required properties:
+- {big,little}-endian: these are boolean properties, if absent
+  meaning that the CPU and the Device are in the same endianness mode,
+  these properties are for register values and all the buffers only.
+
+Examples:
+Scenario 1 : CPU in LE mode & device in LE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+};
+
+Scenario 2 : CPU in LE mode & device in BE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+	      big-endian;
+};
+
+Scenario 3 : CPU in BE mode & device in BE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+};
+
+Scenario 4 : CPU in BE mode & device in LE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+	      little-endian;
+};

Documentation/devicetree/bindings/sound/es8328.txt

+Everest ES8328 audio CODEC
+
+This device supports both I2C and SPI.
+
+Required properties:
+
+  - compatible : "everest,es8328"
+  - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V
+  - AVDD-supply : Regulator providing analog supply voltage 3.3V
+  - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V
+  - IPVDD-supply : Regulator providing analog output voltage 3.3V
+  - clocks : A 22.5792 or 11.2896 MHz clock
+  - reg : the I2C address of the device for I2C, the chip select number for SPI
+
+Pins on the device (for linking into audio routes):
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * LINPUT1
+  * RINPUT1
+  * LINPUT2
+  * RINPUT2
+  * Mic Bias
+
+
+Example:
+
+codec: es8328@11 {
+	compatible = "everest,es8328";
+	DVDD-supply = <&reg_3p3v>;
+	AVDD-supply = <&reg_3p3v>;
+	PVDD-supply = <&reg_3p3v>;
+	HPVDD-supply = <&reg_3p3v>;
+	clocks = <&clks 169>;
+	reg = <0x11>;
+};

Documentation/devicetree/bindings/sound/fsl,esai.txt

@@ -7,7 +7,8 @@ other DSPs. It has up to six transmitters and four receivers.
 
 Required properties:
 
-  - compatible : Compatible list, must contain "fsl,imx35-esai".
+  - compatible : Compatible list, must contain "fsl,imx35-esai" or
+		 "fsl,vf610-esai"
 
   - reg : Offset and length of the register set for the device.
 

Documentation/devicetree/bindings/sound/fsl-asoc-card.txt

+Freescale Generic ASoC Sound Card with ASRC support
+
+The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
+SoCs connecting with external CODECs.
+
+The idea of this generic sound card is a bit like ASoC Simple Card. However,
+for Freescale SoCs (especially those released in recent years), most of them
+have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
+this is a specific feature that might be painstakingly controlled and merged
+into the Simple Card.
+
+So having this generic sound card allows all Freescale SoC users to benefit
+from the simplification of a new card support and the capability of the wide
+sample rates support through ASRC.
+
+Note: The card is initially designed for those sound cards who use I2S and
+      PCM DAI formats. However, it'll be also possible to support those non
+      I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
+      as the driver has been properly upgraded.
+
+
+The compatible list for this generic sound card currently:
+ "fsl,imx-audio-cs42888"
+
+ "fsl,imx-audio-wm8962"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
+
+ "fsl,imx-audio-sgtl5000"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
+
+Required properties:
+
+  - compatible		: Contains one of entries in the compatible list.
+
+  - model		: The user-visible name of this sound complex
+
+  - audio-cpu		: The phandle of an CPU DAI controller
+
+  - audio-codec		: The phandle of an audio codec
+
+  - audio-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. There're a few pre-designed board connectors:
+			   * Line Out Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Mic Jack
+			   * Ext Spk
+			   * AMIC (stands for Analog Microphone Jack)
+			   * DMIC (stands for Digital Microphone Jack)
+
+			  Note: The "Mic Jack" and "AMIC" are redundant while
+			        coexsiting in order to support the old bindings
+				of wm8962 and sgtl5000.
+
+Optional properties:
+
+  - audio-asrc		: The phandle of ASRC. It can be absent if there's no
+			  need to add ASRC support via DPCM.
+
+Example:
+sound-cs42888 {
+	compatible = "fsl,imx-audio-cs42888";
+	model = "cs42888-audio";
+	audio-cpu = <&esai>;
+	audio-asrc = <&asrc>;
+	audio-codec = <&cs42888>;
+	audio-routing =
+		"Line Out Jack", "AOUT1L",
+		"Line Out Jack", "AOUT1R",
+		"Line Out Jack", "AOUT2L",
+		"Line Out Jack", "AOUT2R",
+		"Line Out Jack", "AOUT3L",
+		"Line Out Jack", "AOUT3R",
+		"Line Out Jack", "AOUT4L",
+		"Line Out Jack", "AOUT4R",
+		"AIN1L", "Line In Jack",
+		"AIN1R", "Line In Jack",
+		"AIN2L", "Line In Jack",
+		"AIN2R", "Line In Jack";
+};

Documentation/devicetree/bindings/sound/fsl-sai.txt

@@ -18,9 +18,8 @@ Required properties:
 - pinctrl-names: Must contain a "default" entry.
 - pinctrl-NNN: One property must exist for each entry in pinctrl-names.
   See ../pinctrl/pinctrl-bindings.txt for details of the property values.
-- big-endian-regs: If this property is absent, the little endian mode will
-  be in use as default, or the big endian mode will be in use for all the
-  device registers.
+- big-endian: Boolean property, required if all the FTM_PWM registers
+  are big-endian rather than little-endian.
 - big-endian-data: If this property is absent, the little endian mode will
   be in use as default, or the big endian mode will be in use for all the
   fifo data.
@@ -38,6 +37,6 @@ sai2: sai@40031000 {
 	      dma-names = "tx", "rx";
 	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
 		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
-	      big-endian-regs;
+	      big-endian;
 	      big-endian-data;
 };

Documentation/devicetree/bindings/sound/imx-audio-es8328.txt

+Freescale i.MX audio complex with ES8328 codec
+
+Required properties:
+- compatible       : "fsl,imx-audio-es8328"
+- model            : The user-visible name of this sound complex
+- ssi-controller   : The phandle of the i.MX SSI controller
+- jack-gpio        : Optional GPIO for headphone jack
+- audio-amp-supply : Power regulator for speaker amps
+- audio-codec      : The phandle of the ES8328 audio codec
+- audio-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 could be power supplies, ES8328
+		     pins, and the jacks on the board:
+
+			Power supplies:
+			   * audio-amp
+
+			ES8328 pins:
+			   * LOUT1
+			   * LOUT2
+			   * ROUT1
+			   * ROUT2
+			   * LINPUT1
+			   * LINPUT2
+			   * RINPUT1
+			   * RINPUT2
+			   * Mic PGA
+
+			Board connectors:
+			   * Headphone
+			   * Speaker
+			   * Mic Jack
+- mux-int-port     : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port     : The external port of the i.MX audio muxer (AUDMIX)
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+	compatible = "fsl,imx-audio-es8328";
+	model = "imx-audio-es8328";
+	ssi-controller = <&ssi1>;
+	audio-codec = <&codec>;
+	jack-gpio = <&gpio5 15 0>;
+	audio-amp-supply = <&reg_audio_amp>;
+	audio-routing =
+		"Speaker", "LOUT2",
+		"Speaker", "ROUT2",
+		"Speaker", "audio-amp",
+		"Headphone", "ROUT1",
+		"Headphone", "LOUT1",
+		"LINPUT1", "Mic Jack",
+		"RINPUT1", "Mic Jack",
+		"Mic Jack", "Mic Bias";
+	mux-int-port = <1>;
+	mux-ext-port = <3>;
+};

Documentation/devicetree/bindings/vendor-prefixes.txt

@@ -48,6 +48,7 @@ epfl	Ecole Polytechnique Fédérale de Lausanne
 epson	Seiko Epson Corp.
 est	ESTeem Wireless Modems
 eukrea  Eukréa Electromatique
+everest	Everest Semiconductor Co. Ltd.
 excito	Excito
 fsl	Freescale Semiconductor
 GEFanuc	GE Fanuc Intelligent Platforms Embedded Systems, Inc.

drivers/base/regmap/regmap-i2c.c

@@ -168,6 +168,8 @@ static struct regmap_bus regmap_i2c = {
 	.write = regmap_i2c_write,
 	.gather_write = regmap_i2c_gather_write,
 	.read = regmap_i2c_read,
+	.reg_format_endian_default = REGMAP_ENDIAN_BIG,
+	.val_format_endian_default = REGMAP_ENDIAN_BIG,
 };
 
 static const struct regmap_bus *regmap_get_i2c_bus(struct i2c_client *i2c,

drivers/base/regmap/regmap-spi.c

@@ -109,6 +109,8 @@ static struct regmap_bus regmap_spi = {
 	.async_alloc = regmap_spi_async_alloc,
 	.read = regmap_spi_read,
 	.read_flag_mask = 0x80,
+	.reg_format_endian_default = REGMAP_ENDIAN_BIG,
+	.val_format_endian_default = REGMAP_ENDIAN_BIG,
 };
 
 /**

drivers/base/regmap/regmap.c

@@ -15,6 +15,7 @@
 #include <linux/export.h>
 #include <linux/mutex.h>
 #include <linux/err.h>
+#include <linux/of.h>
 #include <linux/rbtree.h>
 #include <linux/sched.h>
 
@@ -448,6 +449,66 @@ int regmap_attach_dev(struct device *dev, struct regmap *map,
 }
 EXPORT_SYMBOL_GPL(regmap_attach_dev);
 
+static enum regmap_endian regmap_get_reg_endian(const struct regmap_bus *bus,
+					const struct regmap_config *config)
+{
+	enum regmap_endian endian;
+
+	/* Retrieve the endianness specification from the regmap config */
+	endian = config->reg_format_endian;
+
+	/* If the regmap config specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Retrieve the endianness specification from the bus config */
+	if (bus && bus->reg_format_endian_default)
+		endian = bus->reg_format_endian_default;
+
+	/* If the bus specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Use this if no other value was found */
+	return REGMAP_ENDIAN_BIG;
+}
+
+static enum regmap_endian regmap_get_val_endian(struct device *dev,
+					const struct regmap_bus *bus,
+					const struct regmap_config *config)
+{
+	struct device_node *np = dev->of_node;
+	enum regmap_endian endian;
+
+	/* Retrieve the endianness specification from the regmap config */
+	endian = config->val_format_endian;
+
+	/* If the regmap config specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Parse the device's DT node for an endianness specification */
+	if (of_property_read_bool(np, "big-endian"))
+		endian = REGMAP_ENDIAN_BIG;
+	else if (of_property_read_bool(np, "little-endian"))
+		endian = REGMAP_ENDIAN_LITTLE;
+
+	/* If the endianness was specified in DT, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Retrieve the endianness specification from the bus config */
+	if (bus && bus->val_format_endian_default)
+		endian = bus->val_format_endian_default;
+
+	/* If the bus specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Use this if no other value was found */
+	return REGMAP_ENDIAN_BIG;
+}
+
 /**
  * regmap_init(): Initialise register map
  *
@@ -551,17 +612,8 @@ struct regmap *regmap_init(struct device *dev,
 		map->reg_read  = _regmap_bus_read;
 	}
 
-	reg_endian = config->reg_format_endian;
-	if (reg_endian == REGMAP_ENDIAN_DEFAULT)
-		reg_endian = bus->reg_format_endian_default;
-	if (reg_endian == REGMAP_ENDIAN_DEFAULT)
-		reg_endian = REGMAP_ENDIAN_BIG;
-
-	val_endian = config->val_format_endian;
-	if (val_endian == REGMAP_ENDIAN_DEFAULT)
-		val_endian = bus->val_format_endian_default;
-	if (val_endian == REGMAP_ENDIAN_DEFAULT)
-		val_endian = REGMAP_ENDIAN_BIG;
+	reg_endian = regmap_get_reg_endian(bus, config);
+	val_endian = regmap_get_val_endian(dev, bus, config);
 
 	switch (config->reg_bits + map->reg_shift) {
 	case 2:

sound/soc/codecs/Kconfig

@@ -57,6 +57,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
 	select SND_SOC_BT_SCO
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
@@ -405,6 +407,17 @@ config SND_SOC_DMIC
 config SND_SOC_HDMI_CODEC
        tristate "HDMI stub CODEC"
 
+config SND_SOC_ES8328
+	tristate "Everest Semi ES8328 CODEC"
+
+config SND_SOC_ES8328_I2C
+	tristate
+	select SND_SOC_ES8328
+
+config SND_SOC_ES8328_SPI
+	tristate
+	select SND_SOC_ES8328
+
 config SND_SOC_ISABELLE
         tristate
 

sound/soc/codecs/Makefile

@@ -49,6 +49,9 @@ snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
 snd-soc-bt-sco-objs := bt-sco.o
 snd-soc-dmic-objs := dmic.o
+snd-soc-es8328-objs := es8328.o
+snd-soc-es8328-i2c-objs := es8328-i2c.o
+snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
@@ -220,6 +223,9 @@ obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
 obj-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
+obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
+obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o

sound/soc/codecs/es8328-i2c.c

+/*
+ * es8328-i2c.c  --  ES8328 ALSA SoC I2C Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "es8328.h"
+
+static const struct i2c_device_id es8328_id[] = {
+	{ "everest,es8328", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, es8328_id);
+
+static const struct of_device_id es8328_of_match[] = {
+	{ .compatible = "everest,es8328", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es8328_of_match);
+
+static int es8328_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	return es8328_probe(&i2c->dev,
+			devm_regmap_init_i2c(i2c, &es8328_regmap_config));
+}
+
+static int es8328_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_codec(&i2c->dev);
+	return 0;
+}
+
+static struct i2c_driver es8328_i2c_driver = {
+	.driver = {
+		.name		= "es8328",
+		.of_match_table = es8328_of_match,
+	},
+	.probe    = es8328_i2c_probe,
+	.remove   = es8328_i2c_remove,
+	.id_table = es8328_id,
+};
+
+module_i2c_driver(es8328_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ES8328 audio CODEC I2C driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/es8328-spi.c

+/*
+ * es8328.c  --  ES8328 ALSA SoC SPI Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+#include "es8328.h"
+
+static const struct of_device_id es8328_of_match[] = {
+	{ .compatible = "everest,es8328", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es8328_of_match);
+
+static int es8328_spi_probe(struct spi_device *spi)
+{
+	return es8328_probe(&spi->dev,
+			devm_regmap_init_spi(spi, &es8328_regmap_config));
+}
+
+static int es8328_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	return 0;
+}
+
+static struct spi_driver es8328_spi_driver = {
+	.driver = {
+		.name		= "es8328",
+		.of_match_table	= es8328_of_match,
+	},
+	.probe	= es8328_spi_probe,
+	.remove	= es8328_spi_remove,
+};
+
+module_spi_driver(es8328_spi_driver);
+MODULE_DESCRIPTION("ASoC ES8328 audio CODEC SPI driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/es8328.c

+/*
+ * es8328.c  --  ES8328 ALSA SoC Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "es8328.h"
+
+#define ES8328_SYSCLK_RATE_1X 11289600
+#define ES8328_SYSCLK_RATE_2X 22579200
+
+/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
+static struct {
+	int rate;
+	u8 ratio;
+} mclk_ratios[] = {
+	{ 8000, 9 },
+	{11025, 7 },
+	{22050, 4 },
+	{44100, 2 },
+};
+
+/* regulator supplies for sgtl5000, VDDD is an optional external supply */
+enum sgtl5000_regulator_supplies {
+	DVDD,
+	AVDD,
+	PVDD,
+	HPVDD,
+	ES8328_SUPPLY_NUM
+};
+
+/* vddd is optional supply */
+static const char * const supply_names[ES8328_SUPPLY_NUM] = {
+	"DVDD",
+	"AVDD",
+	"PVDD",
+	"HPVDD",
+};
+
+#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_22050 | \
+		SNDRV_PCM_RATE_11025)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+
+struct es8328_priv {
+	struct regmap *regmap;
+	struct clk *clk;
+	int playback_fs;
+	bool deemph;
+	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
+};
+
+/*
+ * ES8328 Controls
+ */
+
+static const char * const adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+					  "L + R Invert"};
+static SOC_ENUM_SINGLE_DECL(adcpol,
+			    ES8328_ADCCONTROL6, 6, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0);
+static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0);
+
+static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+
+static int es8328_set_deemph(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int val, i, best;
+
+	/*
+	 * If we're using deemphasis select the nearest available sample
+	 * rate.
+	 */
+	if (es8328->deemph) {
+		best = 1;
+		for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
+			if (abs(deemph_settings[i] - es8328->playback_fs) <
+			    abs(deemph_settings[best] - es8328->playback_fs))
+				best = i;
+		}
+
+		val = best << 1;
+	} else {
+		val = 0;
+	}
+
+	dev_dbg(codec->dev, "Set deemphasis %d\n", val);
+
+	return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val);
+}
+
+static int es8328_get_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = es8328->deemph;
+	return 0;
+}
+
+static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int deemph = ucontrol->value.enumerated.item[0];
+	int ret;
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	ret = es8328_set_deemph(codec);
+	if (ret < 0)
+		return ret;
+
+	es8328->deemph = deemph;
+
+	return 0;
+}
+
+
+
+static const struct snd_kcontrol_new es8328_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Capture Digital Volume",
+		ES8328_ADCCONTROL8, ES8328_ADCCONTROL9,
+		 0, 0xc0, 1, dac_adc_tlv),
+	SOC_SINGLE("Capture ZC Switch", ES8328_ADCCONTROL7, 6, 1, 0),
+
+	SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
+		    es8328_get_deemph, es8328_put_deemph),
+
+	SOC_ENUM("Capture Polarity", adcpol),
+
+	SOC_SINGLE_TLV("Left Mixer Left Bypass Volume",
+			ES8328_DACCONTROL17, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Left Mixer Right Bypass Volume",
+			ES8328_DACCONTROL19, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Right Mixer Left Bypass Volume",
+			ES8328_DACCONTROL18, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Right Mixer Right Bypass Volume",
+			ES8328_DACCONTROL20, 3, 7, 1, bypass_tlv),
+
+	SOC_DOUBLE_R_TLV("PCM Volume",
+			ES8328_LDACVOL, ES8328_RDACVOL,
+			0, ES8328_DACVOL_MAX, 1, dac_adc_tlv),
+
+	SOC_DOUBLE_R_TLV("Output 1 Playback Volume",
+			ES8328_LOUT1VOL, ES8328_ROUT1VOL,
+			0, ES8328_OUT1VOL_MAX, 0, play_tlv),
+
+	SOC_DOUBLE_R_TLV("Output 2 Playback Volume",
+			ES8328_LOUT2VOL, ES8328_ROUT2VOL,
+			0, ES8328_OUT2VOL_MAX, 0, play_tlv),
+
+	SOC_DOUBLE_TLV("Mic PGA Volume", ES8328_ADCCONTROL1,
+			4, 0, 8, 0, mic_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+
+static const char * const es8328_line_texts[] = {
+	"Line 1", "Line 2", "PGA", "Differential"};
+
+static const struct soc_enum es8328_lline_enum =
+	SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 3,
+			      ARRAY_SIZE(es8328_line_texts),
+			      es8328_line_texts);
+static const struct snd_kcontrol_new es8328_left_line_controls =
+	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+
+static const struct soc_enum es8328_rline_enum =
+	SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 0,
+			      ARRAY_SIZE(es8328_line_texts),
+			      es8328_line_texts);
+static const struct snd_kcontrol_new es8328_right_line_controls =
+	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0),
+};
+
+static const char * const es8328_pga_sel[] = {
+	"Line 1", "Line 2", "Line 3", "Differential"};
+
+/* Left PGA Mux */
+static const struct soc_enum es8328_lpga_enum =
+	SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 6,
+			      ARRAY_SIZE(es8328_pga_sel),
+			      es8328_pga_sel);
+static const struct snd_kcontrol_new es8328_left_pga_controls =
+	SOC_DAPM_ENUM("Route", es8328_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum es8328_rpga_enum =
+	SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 4,
+			      ARRAY_SIZE(es8328_pga_sel),
+			      es8328_pga_sel);
+static const struct snd_kcontrol_new es8328_right_pga_controls =
+	SOC_DAPM_ENUM("Route", es8328_rpga_enum);
+
+/* Differential Mux */
+static const char * const es8328_diff_sel[] = {"Line 1", "Line 2"};
+static SOC_ENUM_SINGLE_DECL(diffmux,
+			    ES8328_ADCCONTROL3, 7, es8328_diff_sel);
+static const struct snd_kcontrol_new es8328_diffmux_controls =
+	SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char * const es8328_mono_mux[] = {"Stereo", "Mono (Left)",
+	"Mono (Right)", "Digital Mono"};
+static SOC_ENUM_SINGLE_DECL(monomux,
+			    ES8328_ADCCONTROL3, 3, es8328_mono_mux);
+static const struct snd_kcontrol_new es8328_monomux_controls =
+	SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_diffmux_controls),
+	SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_monomux_controls),
+	SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_monomux_controls),
+
+	SND_SOC_DAPM_MUX("Left PGA Mux", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_AINL_OFF, 1,
+			&es8328_left_pga_controls),
+	SND_SOC_DAPM_MUX("Right PGA Mux", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_AINR_OFF, 1,
+			&es8328_right_pga_controls),
+
+	SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_left_line_controls),
+	SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_right_line_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADCR_OFF, 1),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADCL_OFF, 1),
+
+	SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC STM", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACSTM_RESET, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC STM", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCSTM_RESET, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC DIG", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACDIG_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC DIG", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCDIG_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC DLL", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACDLL_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC DLL", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCDLL_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("ADC Vref", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCVREF_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC Vref", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACVREF_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", ES8328_DACPOWER,
+			ES8328_DACPOWER_RDAC_OFF, 1),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER,
+			ES8328_DACPOWER_LDAC_OFF, 1),
+
+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+		&es8328_left_mixer_controls[0],
+		ARRAY_SIZE(es8328_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+		&es8328_right_mixer_controls[0],
+		ARRAY_SIZE(es8328_right_mixer_controls)),
+
+	SND_SOC_DAPM_PGA("Right Out 2", ES8328_DACPOWER,
+			ES8328_DACPOWER_ROUT2_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 2", ES8328_DACPOWER,
+			ES8328_DACPOWER_LOUT2_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Out 1", ES8328_DACPOWER,
+			ES8328_DACPOWER_ROUT1_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 1", ES8328_DACPOWER,
+			ES8328_DACPOWER_LOUT1_ON, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+
+	SND_SOC_DAPM_INPUT("LINPUT1"),
+	SND_SOC_DAPM_INPUT("LINPUT2"),
+	SND_SOC_DAPM_INPUT("RINPUT1"),
+	SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route es8328_dapm_routes[] = {
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left PGA Mux", "Line 1", "LINPUT1" },
+	{ "Left PGA Mux", "Line 2", "LINPUT2" },
+	{ "Left PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Right PGA Mux", "Line 1", "RINPUT1" },
+	{ "Right PGA Mux", "Line 2", "RINPUT2" },
+	{ "Right PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Differential Mux", "Line 1", "LINPUT1" },
+	{ "Differential Mux", "Line 1", "RINPUT1" },
+	{ "Differential Mux", "Line 2", "LINPUT2" },
+	{ "Differential Mux", "Line 2", "RINPUT2" },
+
+	{ "Left ADC Mux", "Stereo", "Left PGA Mux" },
+	{ "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+	{ "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+	{ "Right ADC Mux", "Stereo", "Right PGA Mux" },
+	{ "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+	{ "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+	{ "Left ADC", NULL, "Left ADC Mux" },
+	{ "Right ADC", NULL, "Right ADC Mux" },
+
+	{ "ADC DIG", NULL, "ADC STM" },
+	{ "ADC DIG", NULL, "ADC Vref" },
+	{ "ADC DIG", NULL, "ADC DLL" },
+
+	{ "Left ADC", NULL, "ADC DIG" },
+	{ "Right ADC", NULL, "ADC DIG" },
+
+	{ "Mic Bias", NULL, "Mic Bias Gen" },
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left Out 1", NULL, "Left DAC" },
+	{ "Right Out 1", NULL, "Right DAC" },
+	{ "Left Out 2", NULL, "Left DAC" },
+	{ "Right Out 2", NULL, "Right DAC" },
+
+	{ "Left Mixer", "Playback Switch", "Left DAC" },
+	{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Left Mixer", "Right Playback Switch", "Right DAC" },
+	{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "Right Mixer", "Left Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Right Mixer", "Playback Switch", "Right DAC" },
+	{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "DAC DIG", NULL, "DAC STM" },
+	{ "DAC DIG", NULL, "DAC Vref" },
+	{ "DAC DIG", NULL, "DAC DLL" },
+
+	{ "Left DAC", NULL, "DAC DIG" },
+	{ "Right DAC", NULL, "DAC DIG" },
+
+	{ "Left Out 1", NULL, "Left Mixer" },
+	{ "LOUT1", NULL, "Left Out 1" },
+	{ "Right Out 1", NULL, "Right Mixer" },
+	{ "ROUT1", NULL, "Right Out 1" },
+
+	{ "Left Out 2", NULL, "Left Mixer" },
+	{ "LOUT2", NULL, "Left Out 2" },
+	{ "Right Out 2", NULL, "Right Mixer" },
+	{ "ROUT2", NULL, "Right Out 2" },
+};
+
+static int es8328_mute(struct snd_soc_dai *dai, int mute)
+{
+	return snd_soc_update_bits(dai->codec, ES8328_DACCONTROL3,
+			ES8328_DACCONTROL3_DACMUTE,
+			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
+}
+
+static int es8328_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int clk_rate;
+	int i;
+	int reg;
+	u8 ratio;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = ES8328_DACCONTROL2;
+	else
+		reg = ES8328_ADCCONTROL5;
+
+	clk_rate = clk_get_rate(es8328->clk);
+
+	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
+		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
+		dev_err(codec->dev,
+			"%s: clock is running at %d Hz, not %d or %d Hz\n",
+			 __func__, clk_rate,
+			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+		return -EINVAL;
+	}
+
+	/* find master mode MCLK to sampling frequency ratio */
+	ratio = mclk_ratios[0].rate;
+	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
+		if (params_rate(params) <= mclk_ratios[i].rate)
+			ratio = mclk_ratios[i].ratio;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		es8328->playback_fs = params_rate(params);
+		es8328_set_deemph(codec);
+	}
+
+	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
+}
+
+static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int clk_rate;
+	u8 mode = ES8328_DACCONTROL1_DACWL_16;
+
+	/* set master/slave audio interface */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
+		return -EINVAL;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
+		return -EINVAL;
+
+	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
+	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+
+	/* Master serial port mode, with BCLK generated automatically */
+	clk_rate = clk_get_rate(es8328->clk);
+	if (clk_rate == ES8328_SYSCLK_RATE_1X)
+		snd_soc_write(codec, ES8328_MASTERMODE,
+				ES8328_MASTERMODE_MSC);
+	else
+		snd_soc_write(codec, ES8328_MASTERMODE,
+				ES8328_MASTERMODE_MCLKDIV2 |
+				ES8328_MASTERMODE_MSC);
+
+	return 0;
+}
+
+static int es8328_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VREF, VMID=2x50k, digital enabled */
+		snd_soc_write(codec, ES8328_CHIPPOWER, 0);
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				ES8328_CONTROL1_VMIDSEL_50k |
+				ES8328_CONTROL1_ENREF);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_update_bits(codec, ES8328_CONTROL1,
+					ES8328_CONTROL1_VMIDSEL_MASK |
+					ES8328_CONTROL1_ENREF,
+					ES8328_CONTROL1_VMIDSEL_5k |
+					ES8328_CONTROL1_ENREF);
+
+			/* Charge caps */
+			msleep(100);
+		}
+
+		snd_soc_write(codec, ES8328_CONTROL2,
+				ES8328_CONTROL2_OVERCURRENT_ON |
+				ES8328_CONTROL2_THERMAL_SHUTDOWN_ON);
+
+		/* VREF, VMID=2*500k, digital stopped */
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				ES8328_CONTROL1_VMIDSEL_500k |
+				ES8328_CONTROL1_ENREF);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.hw_params	= es8328_hw_params,
+	.digital_mute	= es8328_mute,
+	.set_fmt	= es8328_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver es8328_dai = {
+	.name = "es8328-hifi-analog",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = ES8328_RATES,
+		.formats = ES8328_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = ES8328_RATES,
+		.formats = ES8328_FORMATS,
+	},
+	.ops = &es8328_dai_ops,
+};
+
+static int es8328_suspend(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	es8328_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	clk_disable_unprepare(es8328->clk);
+
+	ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to disable regulators\n");
+		return ret;
+	}
+	return 0;
+}
+
+static int es8328_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ret = clk_prepare_enable(es8328->clk);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable clock\n");
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies),
+					es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable regulators\n");
+		return ret;
+	}
+
+	regcache_mark_dirty(regmap);
+	ret = regcache_sync(regmap);
+	if (ret) {
+		dev_err(codec->dev, "unable to sync regcache\n");
+		return ret;
+	}
+
+	es8328_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int es8328_codec_probe(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies),
+					es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable regulators\n");
+		return ret;
+	}
+
+	/* Setup clocks */
+	es8328->clk = devm_clk_get(codec->dev, NULL);
+	if (IS_ERR(es8328->clk)) {
+		dev_err(codec->dev, "codec clock missing or invalid\n");
+		goto clk_fail;
+	}
+
+	ret = clk_prepare_enable(es8328->clk);
+	if (ret) {
+		dev_err(codec->dev, "unable to prepare codec clk\n");
+		goto clk_fail;
+	}
+
+	return 0;
+
+clk_fail:
+	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			       es8328->supplies);
+	return ret;
+}
+
+static int es8328_remove(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->clk)
+		clk_disable_unprepare(es8328->clk);
+
+	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			       es8328->supplies);
+
+	return 0;
+}
+
+const struct regmap_config es8328_regmap_config = {
+	.reg_bits	= 8,
+	.val_bits	= 8,
+	.max_register	= ES8328_REG_MAX,
+	.cache_type	= REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(es8328_regmap_config);
+
+static struct snd_soc_codec_driver es8328_codec_driver = {
+	.probe		  = es8328_codec_probe,
+	.suspend	  = es8328_suspend,
+	.resume		  = es8328_resume,
+	.remove		  = es8328_remove,
+	.set_bias_level	  = es8328_set_bias_level,
+	.controls	  = es8328_snd_controls,
+	.num_controls	  = ARRAY_SIZE(es8328_snd_controls),
+	.dapm_widgets	  = es8328_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(es8328_dapm_widgets),
+	.dapm_routes	  = es8328_dapm_routes,
+	.num_dapm_routes  = ARRAY_SIZE(es8328_dapm_routes),
+};
+
+int es8328_probe(struct device *dev, struct regmap *regmap)
+{
+	struct es8328_priv *es8328;
+	int ret;
+	int i;
+
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	es8328 = devm_kzalloc(dev, sizeof(*es8328), GFP_KERNEL);
+	if (es8328 == NULL)
+		return -ENOMEM;
+
+	es8328->regmap = regmap;
+
+	for (i = 0; i < ARRAY_SIZE(es8328->supplies); i++)
+		es8328->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(es8328->supplies),
+				es8328->supplies);
+	if (ret) {
+		dev_err(dev, "unable to get regulators\n");
+		return ret;
+	}
+
+	dev_set_drvdata(dev, es8328);
+
+	return snd_soc_register_codec(dev,
+			&es8328_codec_driver, &es8328_dai, 1);
+}
+EXPORT_SYMBOL_GPL(es8328_probe);
+
+MODULE_DESCRIPTION("ASoC ES8328 driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/es8328.h

+/*
+ * es8328.h  --  ES8328 ALSA SoC Audio driver
+ */
+
+#ifndef _ES8328_H
+#define _ES8328_H
+
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config es8328_regmap_config;
+int es8328_probe(struct device *dev, struct regmap *regmap);
+
+#define ES8328_DACLVOL 46
+#define ES8328_DACRVOL 47
+#define ES8328_DACCTL 28
+#define ES8328_RATEMASK (0x1f << 0)
+
+#define ES8328_CONTROL1		0x00
+#define ES8328_CONTROL1_VMIDSEL_OFF (0 << 0)
+#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
+#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
+#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_ENREF (1 << 2)
+#define ES8328_CONTROL1_SEQEN (1 << 3)
+#define ES8328_CONTROL1_SAMEFS (1 << 4)
+#define ES8328_CONTROL1_DACMCLK_ADC (0 << 5)
+#define ES8328_CONTROL1_DACMCLK_DAC (1 << 5)
+#define ES8328_CONTROL1_LRCM (1 << 6)
+#define ES8328_CONTROL1_SCP_RESET (1 << 7)
+
+#define ES8328_CONTROL2		0x01
+#define ES8328_CONTROL2_VREF_BUF_OFF (1 << 0)
+#define ES8328_CONTROL2_VREF_LOWPOWER (1 << 1)
+#define ES8328_CONTROL2_IBIASGEN_OFF (1 << 2)
+#define ES8328_CONTROL2_ANALOG_OFF (1 << 3)
+#define ES8328_CONTROL2_VREF_BUF_LOWPOWER (1 << 4)
+#define ES8328_CONTROL2_VCM_MOD_LOWPOWER (1 << 5)
+#define ES8328_CONTROL2_OVERCURRENT_ON (1 << 6)
+#define ES8328_CONTROL2_THERMAL_SHUTDOWN_ON (1 << 7)
+
+#define ES8328_CHIPPOWER	0x02
+#define ES8328_CHIPPOWER_DACVREF_OFF 0
+#define ES8328_CHIPPOWER_ADCVREF_OFF 1
+#define ES8328_CHIPPOWER_DACDLL_OFF 2
+#define ES8328_CHIPPOWER_ADCDLL_OFF 3
+#define ES8328_CHIPPOWER_DACSTM_RESET 4
+#define ES8328_CHIPPOWER_ADCSTM_RESET 5
+#define ES8328_CHIPPOWER_DACDIG_OFF 6
+#define ES8328_CHIPPOWER_ADCDIG_OFF 7
+
+#define ES8328_ADCPOWER		0x03
+#define ES8328_ADCPOWER_INT1_LOWPOWER 0
+#define ES8328_ADCPOWER_FLASH_ADC_LOWPOWER 1
+#define ES8328_ADCPOWER_ADC_BIAS_GEN_OFF 2
+#define ES8328_ADCPOWER_MIC_BIAS_OFF 3
+#define ES8328_ADCPOWER_ADCR_OFF 4
+#define ES8328_ADCPOWER_ADCL_OFF 5
+#define ES8328_ADCPOWER_AINR_OFF 6
+#define ES8328_ADCPOWER_AINL_OFF 7
+
+#define ES8328_DACPOWER		0x04
+#define ES8328_DACPOWER_OUT3_ON 0
+#define ES8328_DACPOWER_MONO_ON 1
+#define ES8328_DACPOWER_ROUT2_ON 2
+#define ES8328_DACPOWER_LOUT2_ON 3
+#define ES8328_DACPOWER_ROUT1_ON 4
+#define ES8328_DACPOWER_LOUT1_ON 5
+#define ES8328_DACPOWER_RDAC_OFF 6
+#define ES8328_DACPOWER_LDAC_OFF 7
+
+#define ES8328_CHIPLOPOW1	0x05
+#define ES8328_CHIPLOPOW2	0x06
+#define ES8328_ANAVOLMANAG	0x07
+
+#define ES8328_MASTERMODE	0x08
+#define ES8328_MASTERMODE_BCLKDIV (0 << 0)
+#define ES8328_MASTERMODE_BCLK_INV (1 << 5)
+#define ES8328_MASTERMODE_MCLKDIV2 (1 << 6)
+#define ES8328_MASTERMODE_MSC (1 << 7)
+
+#define ES8328_ADCCONTROL1	0x09
+#define ES8328_ADCCONTROL2	0x0a
+#define ES8328_ADCCONTROL3	0x0b
+#define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL5	0x0d
+#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
+
+#define ES8328_ADCCONTROL6	0x0e
+
+#define ES8328_ADCCONTROL7	0x0f
+#define ES8328_ADCCONTROL7_ADC_MUTE (1 << 2)
+#define ES8328_ADCCONTROL7_ADC_LER (1 << 3)
+#define ES8328_ADCCONTROL7_ADC_ZERO_CROSS (1 << 4)
+#define ES8328_ADCCONTROL7_ADC_SOFT_RAMP (1 << 5)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_4 (0 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_8 (1 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_16 (2 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_32 (3 << 6)
+
+#define ES8328_ADCCONTROL8	0x10
+#define ES8328_ADCCONTROL9	0x11
+#define ES8328_ADCCONTROL10	0x12
+#define ES8328_ADCCONTROL11	0x13
+#define ES8328_ADCCONTROL12	0x14
+#define ES8328_ADCCONTROL13	0x15
+#define ES8328_ADCCONTROL14	0x16
+
+#define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
+#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
+#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
+#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
+#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
+#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
+#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
+#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
+#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK1 (1 << 6)
+#define ES8328_DACCONTROL1_LRSWAP (1 << 7)
+
+#define ES8328_DACCONTROL2	0x18
+#define ES8328_DACCONTROL2_RATEMASK (0x1f << 0)
+#define ES8328_DACCONTROL2_DOUBLESPEED (1 << 5)
+
+#define ES8328_DACCONTROL3	0x19
+#define ES8328_DACCONTROL3_AUTOMUTE (1 << 2)
+#define ES8328_DACCONTROL3_DACMUTE (1 << 2)
+#define ES8328_DACCONTROL3_LEFTGAINVOL (1 << 3)
+#define ES8328_DACCONTROL3_DACZEROCROSS (1 << 4)
+#define ES8328_DACCONTROL3_DACSOFTRAMP (1 << 5)
+#define ES8328_DACCONTROL3_DACRAMPRATE (3 << 6)
+
+#define ES8328_LDACVOL 0x1a
+#define ES8328_LDACVOL_MASK (0 << 0)
+#define ES8328_LDACVOL_MAX (0xc0)
+
+#define ES8328_RDACVOL 0x1b
+#define ES8328_RDACVOL_MASK (0 << 0)
+#define ES8328_RDACVOL_MAX (0xc0)
+
+#define ES8328_DACVOL_MAX (0xc0)
+
+#define ES8328_DACCONTROL4	0x1a
+#define ES8328_DACCONTROL5	0x1b
+
+#define ES8328_DACCONTROL6	0x1c
+#define ES8328_DACCONTROL6_CLICKFREE (1 << 3)
+#define ES8328_DACCONTROL6_DAC_INVR (1 << 4)
+#define ES8328_DACCONTROL6_DAC_INVL (1 << 5)
+#define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_48k (3 << 6)
+
+#define ES8328_DACCONTROL7	0x1d
+#define ES8328_DACCONTROL7_VPP_SCALE_3p5	(0 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_4p0	(1 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_3p0	(2 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_2p5	(3 << 0)
+#define ES8328_DACCONTROL7_SHELVING_STRENGTH (1 << 2) /* In eights */
+#define ES8328_DACCONTROL7_MONO		(1 << 5)
+#define ES8328_DACCONTROL7_ZEROR	(1 << 6)
+#define ES8328_DACCONTROL7_ZEROL	(1 << 7)
+
+/* Shelving filter */
+#define ES8328_DACCONTROL8	0x1e
+#define ES8328_DACCONTROL9	0x1f
+#define ES8328_DACCONTROL10	0x20
+#define ES8328_DACCONTROL11	0x21
+#define ES8328_DACCONTROL12	0x22
+#define ES8328_DACCONTROL13	0x23
+#define ES8328_DACCONTROL14	0x24
+#define ES8328_DACCONTROL15	0x25
+
+#define ES8328_DACCONTROL16	0x26
+#define ES8328_DACCONTROL16_RMIXSEL_RIN1 (0 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RIN2 (1 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RIN3 (2 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RADC (3 << 0)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN1 (0 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN2 (1 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN3 (2 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LADC (3 << 3)
+
+#define ES8328_DACCONTROL17	0x27
+#define ES8328_DACCONTROL17_LI2LOVOL (7 << 3)
+#define ES8328_DACCONTROL17_LI2LO (1 << 6)
+#define ES8328_DACCONTROL17_LD2LO (1 << 7)
+
+#define ES8328_DACCONTROL18	0x28
+#define ES8328_DACCONTROL18_RI2LOVOL (7 << 3)
+#define ES8328_DACCONTROL18_RI2LO (1 << 6)
+#define ES8328_DACCONTROL18_RD2LO (1 << 7)
+
+#define ES8328_DACCONTROL19	0x29
+#define ES8328_DACCONTROL19_LI2ROVOL (7 << 3)
+#define ES8328_DACCONTROL19_LI2RO (1 << 6)
+#define ES8328_DACCONTROL19_LD2RO (1 << 7)
+
+#define ES8328_DACCONTROL20	0x2a
+#define ES8328_DACCONTROL20_RI2ROVOL (7 << 3)
+#define ES8328_DACCONTROL20_RI2RO (1 << 6)
+#define ES8328_DACCONTROL20_RD2RO (1 << 7)
+
+#define ES8328_DACCONTROL21	0x2b
+#define ES8328_DACCONTROL21_LI2MOVOL (7 << 3)
+#define ES8328_DACCONTROL21_LI2MO (1 << 6)
+#define ES8328_DACCONTROL21_LD2MO (1 << 7)
+
+#define ES8328_DACCONTROL22	0x2c
+#define ES8328_DACCONTROL22_RI2MOVOL (7 << 3)
+#define ES8328_DACCONTROL22_RI2MO (1 << 6)
+#define ES8328_DACCONTROL22_RD2MO (1 << 7)
+
+#define ES8328_DACCONTROL23	0x2d
+#define ES8328_DACCONTROL23_MOUTINV		(1 << 1)
+#define ES8328_DACCONTROL23_HPSWPOL		(1 << 2)
+#define ES8328_DACCONTROL23_HPSWEN		(1 << 3)
+#define ES8328_DACCONTROL23_VROI_1p5k		(0 << 4)
+#define ES8328_DACCONTROL23_VROI_40k		(1 << 4)
+#define ES8328_DACCONTROL23_OUT3_VREF		(0 << 5)
+#define ES8328_DACCONTROL23_OUT3_ROUT1		(1 << 5)
+#define ES8328_DACCONTROL23_OUT3_MONOOUT	(2 << 5)
+#define ES8328_DACCONTROL23_OUT3_RIGHT_MIXER	(3 << 5)
+#define ES8328_DACCONTROL23_ROUT2INV		(1 << 7)
+
+/* LOUT1 Amplifier */
+#define ES8328_LOUT1VOL 0x2e
+#define ES8328_LOUT1VOL_MASK (0 << 5)
+#define ES8328_LOUT1VOL_MAX (0x24)
+
+/* ROUT1 Amplifier */
+#define ES8328_ROUT1VOL 0x2f
+#define ES8328_ROUT1VOL_MASK (0 << 5)
+#define ES8328_ROUT1VOL_MAX (0x24)
+
+#define ES8328_OUT1VOL_MAX (0x24)
+
+/* LOUT2 Amplifier */
+#define ES8328_LOUT2VOL 0x30
+#define ES8328_LOUT2VOL_MASK (0 << 5)
+#define ES8328_LOUT2VOL_MAX (0x24)
+
+/* ROUT2 Amplifier */
+#define ES8328_ROUT2VOL 0x31
+#define ES8328_ROUT2VOL_MASK (0 << 5)
+#define ES8328_ROUT2VOL_MAX (0x24)
+
+#define ES8328_OUT2VOL_MAX (0x24)
+
+/* Mono Out Amplifier */
+#define ES8328_MONOOUTVOL 0x32
+#define ES8328_MONOOUTVOL_MASK (0 << 5)
+#define ES8328_MONOOUTVOL_MAX (0x24)
+
+#define ES8328_DACCONTROL29	0x33
+#define ES8328_DACCONTROL30	0x34
+
+#define ES8328_SYSCLK		0
+
+#define ES8328_REG_MAX		0x35
+
+#define ES8328_PLL1		0
+#define ES8328_PLL2		1
+
+/* clock inputs */
+#define ES8328_MCLK		0
+#define ES8328_PCMCLK		1
+
+/* clock divider id's */
+#define ES8328_PCMDIV		0
+#define ES8328_BCLKDIV		1
+#define ES8328_VXCLKDIV		2
+
+/* PCM clock dividers */
+#define ES8328_PCM_DIV_1	(0 << 6)
+#define ES8328_PCM_DIV_3	(2 << 6)
+#define ES8328_PCM_DIV_5_5	(3 << 6)
+#define ES8328_PCM_DIV_2	(4 << 6)
+#define ES8328_PCM_DIV_4	(5 << 6)
+#define ES8328_PCM_DIV_6	(6 << 6)
+#define ES8328_PCM_DIV_8	(7 << 6)
+
+/* BCLK clock dividers */
+#define ES8328_BCLK_DIV_1	(0 << 7)
+#define ES8328_BCLK_DIV_2	(1 << 7)
+#define ES8328_BCLK_DIV_4	(2 << 7)
+#define ES8328_BCLK_DIV_8	(3 << 7)
+
+/* VXCLK clock dividers */
+#define ES8328_VXCLK_DIV_1	(0 << 6)
+#define ES8328_VXCLK_DIV_2	(1 << 6)
+#define ES8328_VXCLK_DIV_4	(2 << 6)
+#define ES8328_VXCLK_DIV_8	(3 << 6)
+#define ES8328_VXCLK_DIV_16	(4 << 6)
+
+#define ES8328_DAI_HIFI		0
+#define ES8328_DAI_VOICE	1
+
+#define ES8328_1536FS		1536
+#define ES8328_1024FS		1024
+#define ES8328_768FS		768
+#define ES8328_512FS		512
+#define ES8328_384FS		384
+#define ES8328_256FS		256
+#define ES8328_128FS		128
+
+#endif

sound/soc/fsl/Kconfig

@@ -240,6 +240,18 @@ config SND_SOC_IMX_WM8962
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a wm8962 codec.
 
+config SND_SOC_IMX_ES8328
+	tristate "SoC Audio support for i.MX boards with the ES8328 codec"
+	depends on OF && (I2C || SPI)
+	select SND_SOC_ES8328_I2C if I2C
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_FSL_SSI
+	help
+	  Say Y if you want to add support for the ES8328 audio codec connected
+	  via SSI/I2S over either SPI or I2C.
+
 config SND_SOC_IMX_SGTL5000
 	tristate "SoC Audio support for i.MX boards with sgtl5000"
 	depends on OF && I2C
@@ -268,6 +280,23 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
+config SND_SOC_FSL_ASOC_CARD
+	tristate "Generic ASoC Sound Card with ASRC support"
+	depends on OF && I2C
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_ESAI
+	select SND_SOC_FSL_SAI
+	select SND_SOC_FSL_SSI
+	select SND_SOC_CS42XX8_I2C
+	select SND_SOC_SGTL5000
+	select SND_SOC_WM8962
+	help
+	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
+	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
+	 and SGTL5000.
+	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+
 endif # SND_IMX_SOC
 
 endmenu

sound/soc/fsl/Makefile

@@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
 # Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
 snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
 snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-y := fsl_ssi.o
@@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
 obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
@@ -50,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
 snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-es8328-objs := imx-es8328.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
 snd-soc-imx-spdif-objs := imx-spdif.o
@@ -59,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
 obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
 obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o

sound/soc/fsl/fsl-asoc-card.c

+/*
+ * Freescale Generic ASoC Sound Card driver with ASRC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "fsl_esai.h"
+#include "fsl_sai.h"
+#include "imx-audmux.h"
+
+#include "../codecs/sgtl5000.h"
+#include "../codecs/wm8962.h"
+
+#define RX 0
+#define TX 1
+
+/* Default DAI format without Master and Slave flag */
+#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
+
+/**
+ * CODEC private data
+ *
+ * @mclk_freq: Clock rate of MCLK
+ * @mclk_id: MCLK (or main clock) id for set_sysclk()
+ * @fll_id: FLL (or secordary clock) id for set_sysclk()
+ * @pll_id: PLL id for set_pll()
+ */
+struct codec_priv {
+	unsigned long mclk_freq;
+	u32 mclk_id;
+	u32 fll_id;
+	u32 pll_id;
+};
+
+/**
+ * CPU private data
+ *
+ * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
+ * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
+ * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ *
+ * Note: [1] for tx and [0] for rx
+ */
+struct cpu_priv {
+	unsigned long sysclk_freq[2];
+	u32 sysclk_dir[2];
+	u32 sysclk_id[2];
+};
+
+/**
+ * Freescale Generic ASOC card private data
+ *
+ * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * @pdev: platform device pointer
+ * @codec_priv: CODEC private data
+ * @cpu_priv: CPU private data
+ * @card: ASoC card structure
+ * @sample_rate: Current sample rate
+ * @sample_format: Current sample format
+ * @asrc_rate: ASRC sample rate used by Back-Ends
+ * @asrc_format: ASRC sample format used by Back-Ends
+ * @dai_fmt: DAI format between CPU and CODEC
+ * @name: Card name
+ */
+
+struct fsl_asoc_card_priv {
+	struct snd_soc_dai_link dai_link[3];
+	struct platform_device *pdev;
+	struct codec_priv codec_priv;
+	struct cpu_priv cpu_priv;
+	struct snd_soc_card card;
+	u32 sample_rate;
+	u32 sample_format;
+	u32 asrc_rate;
+	u32 asrc_format;
+	u32 dai_fmt;
+	char name[32];
+};
+
+/**
+ * This dapm route map exsits for DPCM link only.
+ * The other routes shall go through Device Tree.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+	{"Playback",  NULL, "CPU-Playback"},
+	{"ASRC-Capture",  NULL, "CPU-Capture"},
+	{"CPU-Capture",  NULL, "Capture"},
+};
+
+/* Add all possible widgets into here without being redundant */
+static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("AMIC", NULL),
+	SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	struct cpu_priv *cpu_priv = &priv->cpu_priv;
+	struct device *dev = rtd->card->dev;
+	int ret;
+
+	priv->sample_rate = params_rate(params);
+	priv->sample_format = params_format(params);
+
+	if (priv->card.set_bias_level)
+		return 0;
+
+	/* Specific configurations of DAIs starts from here */
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+				     cpu_priv->sysclk_freq[tx],
+				     cpu_priv->sysclk_dir[tx]);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk for cpu dai\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops fsl_asoc_card_ops = {
+	.hw_params = fsl_asoc_card_hw_params,
+};
+
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+			      struct snd_pcm_hw_params *params)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_interval *rate;
+	struct snd_mask *mask;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	rate->max = rate->min = priv->asrc_rate;
+
+	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	snd_mask_none(mask);
+	snd_mask_set(mask, priv->asrc_format);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
+	/* Default ASoC DAI Link*/
+	{
+		.name = "HiFi",
+		.stream_name = "HiFi",
+		.ops = &fsl_asoc_card_ops,
+	},
+	/* DPCM Link between Front-End and Back-End (Optional) */
+	{
+		.name = "HiFi-ASRC-FE",
+		.stream_name = "HiFi-ASRC-FE",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.dynamic = 1,
+	},
+	{
+		.name = "HiFi-ASRC-BE",
+		.stream_name = "HiFi-ASRC-BE",
+		.platform_name = "snd-soc-dummy",
+		.be_hw_params_fixup = be_hw_params_fixup,
+		.ops = &fsl_asoc_card_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	unsigned int pll_out;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
+			break;
+
+		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+			pll_out = priv->sample_rate * 384;
+		else
+			pll_out = priv->sample_rate * 256;
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
+					  codec_priv->mclk_id,
+					  codec_priv->mclk_freq, pll_out);
+		if (ret) {
+			dev_err(dev, "failed to start FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
+					     pll_out, SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
+			break;
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+					     codec_priv->mclk_freq,
+					     SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
+		if (ret) {
+			dev_err(dev, "failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_audmux_init(struct device_node *np,
+				     struct fsl_asoc_card_priv *priv)
+{
+	struct device *dev = &priv->pdev->dev;
+	u32 int_ptcr = 0, ext_ptcr = 0;
+	int int_port, ext_port;
+	int ret;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		return ret;
+	}
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		return ret;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the AUDMUX API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+
+	/*
+	 * Use asynchronous mode (6 wires) for all cases.
+	 * If only 4 wires are needed, just set SSI into
+	 * synchronous mode and enable 4 PADs in IOMUX.
+	 */
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Asynchronous mode can not be set along with RCLKDIR */
+	ret = imx_audmux_v2_configure_port(int_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk in %s\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_probe(struct platform_device *pdev)
+{
+	struct device_node *cpu_np, *codec_np, *asrc_np;
+	struct device_node *np = pdev->dev.of_node;
+	struct platform_device *asrc_pdev = NULL;
+	struct platform_device *cpu_pdev;
+	struct fsl_asoc_card_priv *priv;
+	struct i2c_client *codec_dev;
+	struct clk *codec_clk;
+	u32 width;
+	int ret;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
+	/* Give a chance to old DT binding */
+	if (!cpu_np)
+		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!cpu_np || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	cpu_pdev = of_find_device_by_node(cpu_np);
+	if (!cpu_pdev) {
+		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	codec_dev = of_find_i2c_device_by_node(codec_np);
+	if (!codec_dev) {
+		dev_err(&pdev->dev, "failed to find codec platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
+	if (asrc_np)
+		asrc_pdev = of_find_device_by_node(asrc_np);
+
+	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
+	codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (!IS_ERR(codec_clk)) {
+		priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+		clk_put(codec_clk);
+	}
+
+	/* Default sample rate and format, will be updated in hw_params() */
+	priv->sample_rate = 44100;
+	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+	/* Assign a default DAI format, and allow each card to overwrite it */
+	priv->dai_fmt = DAI_FMT_BASE;
+
+	/* Diversify the card configurations */
+	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
+		priv->card.set_bias_level = NULL;
+		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
+		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
+		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
+		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
+		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
+		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
+		priv->codec_priv.pll_id = WM8962_FLL;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else {
+		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
+		return -EINVAL;
+	}
+
+	/* Common settings for corresponding Freescale CPU DAI driver */
+	if (strstr(cpu_np->name, "ssi")) {
+		/* Only SSI needs to configure AUDMUX */
+		ret = fsl_asoc_card_audmux_init(np, priv);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to init audmux\n");
+			goto asrc_fail;
+		}
+	} else if (strstr(cpu_np->name, "esai")) {
+		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
+		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
+	} else if (strstr(cpu_np->name, "sai")) {
+		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
+		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
+	}
+
+	sprintf(priv->name, "%s-audio", codec_dev->name);
+
+	/* Initialize sound card */
+	priv->pdev = pdev;
+	priv->card.dev = &pdev->dev;
+	priv->card.name = priv->name;
+	priv->card.dai_link = priv->dai_link;
+	priv->card.dapm_routes = audio_map;
+	priv->card.late_probe = fsl_asoc_card_late_probe;
+	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
+	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
+	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
+
+	memcpy(priv->dai_link, fsl_asoc_card_dai,
+	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+	/* Normal DAI Link */
+	priv->dai_link[0].cpu_of_node = cpu_np;
+	priv->dai_link[0].codec_of_node = codec_np;
+	priv->dai_link[0].codec_dai_name = codec_dev->name;
+	priv->dai_link[0].platform_of_node = cpu_np;
+	priv->dai_link[0].dai_fmt = priv->dai_fmt;
+	priv->card.num_links = 1;
+
+	if (asrc_pdev) {
+		/* DPCM DAI Links only if ASRC exsits */
+		priv->dai_link[1].cpu_of_node = asrc_np;
+		priv->dai_link[1].platform_of_node = asrc_np;
+		priv->dai_link[2].codec_dai_name = codec_dev->name;
+		priv->dai_link[2].codec_of_node = codec_np;
+		priv->dai_link[2].cpu_of_node = cpu_np;
+		priv->dai_link[2].dai_fmt = priv->dai_fmt;
+		priv->card.num_links = 3;
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
+					   &priv->asrc_rate);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto asrc_fail;
+		}
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto asrc_fail;
+		}
+
+		if (width == 24)
+			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
+		else
+			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
+	}
+
+	/* Finish card registering */
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+asrc_fail:
+	of_node_put(asrc_np);
+fail:
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	return ret;
+}
+
+static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-cs42888", },
+	{ .compatible = "fsl,imx-audio-sgtl5000", },
+	{ .compatible = "fsl,imx-audio-wm8962", },
+	{}
+};
+
+static struct platform_driver fsl_asoc_card_driver = {
+	.probe = fsl_asoc_card_probe,
+	.driver = {
+		.name = "fsl-asoc-card",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = fsl_asoc_card_dt_ids,
+	},
+};
+module_platform_driver(fsl_asoc_card_driver);
+
+MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asoc-card");
+MODULE_LICENSE("GPL");

sound/soc/fsl/fsl_asrc.c

@@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_asrc_regmap_config = {
+static const struct regmap_config fsl_asrc_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 
 	asrc_priv->paddr = res->start;
 
-	/* Register regmap and let it prepare core clock */
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
 						      &fsl_asrc_regmap_config);
 	if (IS_ERR(asrc_priv->regmap)) {

sound/soc/fsl/fsl_esai.c

@@ -37,6 +37,7 @@
  * @fsysclk: system clock source to derive HCK, SCK and FS
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
+ * @slots: number of slots
  * @hck_rate: clock rate of desired HCKx clock
  * @sck_rate: clock rate of desired SCKx clock
  * @hck_dir: the direction of HCKx pads
@@ -55,6 +56,7 @@ struct fsl_esai {
 	struct clk *fsysclk;
 	u32 fifo_depth;
 	u32 slot_width;
+	u32 slots;
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -362,6 +364,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
 
 	esai_priv->slot_width = slot_width;
+	esai_priv->slots = slots;
 
 	return 0;
 }
@@ -509,10 +512,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 width = snd_pcm_format_width(params_format(params));
 	u32 channels = params_channels(params);
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 	u32 bclk, mask, val;
 	int ret;
 
-	bclk = params_rate(params) * esai_priv->slot_width * 2;
+	bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
 
 	ret = fsl_esai_set_bclk(dai, tx, bclk);
 	if (ret)
@@ -529,7 +533,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
 	      (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
 	val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
-	     (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
+	     (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins));
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
 
@@ -564,6 +568,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u8 i, channels = substream->runtime->channels;
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -578,7 +583,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
-				   tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels));
+				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -705,7 +710,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_esai_regmap_config = {
+static const struct regmap_config fsl_esai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -731,9 +736,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	esai_priv->pdev = pdev;
 	strcpy(esai_priv->name, np->name);
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
@@ -781,6 +783,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	/* Set a default slot size */
 	esai_priv->slot_width = 32;
 
+	/* Set a default slot number */
+	esai_priv->slots = 2;
+
 	/* Set a default master/slave state */
 	esai_priv->slave_mode = true;
 

sound/soc/fsl/fsl_esai.h

@@ -130,8 +130,8 @@
 #define ESAI_xFCR_RE_WIDTH	4
 #define ESAI_xFCR_TE_MASK	(((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
 #define ESAI_xFCR_RE_MASK	(((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
-#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK)
-#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK)
 #define ESAI_xFCR_xFR_SHIFT	1
 #define ESAI_xFCR_xFR_MASK	(1 << ESAI_xFCR_xFR_SHIFT)
 #define ESAI_xFCR_xFR		(1 << ESAI_xFCR_xFR_SHIFT)
@@ -272,8 +272,8 @@
 #define ESAI_xCR_RE_WIDTH	4
 #define ESAI_xCR_TE_MASK	(((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
 #define ESAI_xCR_RE_MASK	(((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
-#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK)
-#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK)
+#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK)
 
 /*
  * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8

sound/soc/fsl/fsl_sai.c

@@ -539,7 +539,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_sai_regmap_config = {
+static const struct regmap_config fsl_sai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -568,10 +568,6 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
 		sai->sai_on_imx = true;
 
-	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
-	if (sai->big_endian_regs)
-		fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);

sound/soc/fsl/fsl_sai.h

@@ -131,7 +131,6 @@ struct fsl_sai {
 	struct clk *bus_clk;
 	struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
 
-	bool big_endian_regs;
 	bool big_endian_data;
 	bool is_dsp_mode;
 	bool sai_on_imx;

sound/soc/fsl/fsl_spdif.c

@@ -1040,7 +1040,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_spdif_regmap_config = {
+static const struct regmap_config fsl_spdif_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1184,9 +1184,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
 	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);

sound/soc/fsl/imx-es8328.c

+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE	32
+#define MUX_PORT_MAX	7
+
+struct imx_es8328_data {
+	struct device *dev;
+	struct snd_soc_dai_link dai;
+	struct snd_soc_card card;
+	char codec_dai_name[DAI_NAME_SIZE];
+	char platform_name[DAI_NAME_SIZE];
+	int jack_gpio;
+};
+
+static struct snd_soc_jack_gpio headset_jack_gpios[] = {
+	{
+		.gpio = -1,
+		.name = "headset-gpio",
+		.report = SND_JACK_HEADSET,
+		.invert = 0,
+		.debounce_time = 200,
+	},
+};
+
+static struct snd_soc_jack headset_jack;
+
+static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct imx_es8328_data *data = container_of(rtd->card,
+					struct imx_es8328_data, card);
+	int ret = 0;
+
+	/* Headphone jack detection */
+	if (gpio_is_valid(data->jack_gpio)) {
+		ret = snd_soc_jack_new(rtd->codec, "Headphone",
+				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+				       &headset_jack);
+		if (ret)
+			return ret;
+
+		headset_jack_gpios[0].gpio = data->jack_gpio;
+		ret = snd_soc_jack_add_gpios(&headset_jack,
+					     ARRAY_SIZE(headset_jack_gpios),
+					     headset_jack_gpios);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0),
+};
+
+static int imx_es8328_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *ssi_np, *codec_np;
+	struct platform_device *ssi_pdev;
+	struct imx_es8328_data *data;
+	u32 int_port, ext_port;
+	int ret;
+	struct device *dev = &pdev->dev;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		goto fail;
+	}
+	if (int_port > MUX_PORT_MAX || int_port == 0) {
+		dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		goto fail;
+	}
+	if (ext_port > MUX_PORT_MAX || ext_port == 0) {
+		dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the audmux API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+	ret = imx_audmux_v2_configure_port(int_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+	ret = imx_audmux_v2_configure_port(ext_port,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+	if (!ssi_np || !codec_np) {
+		dev_err(dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ssi_pdev = of_find_device_by_node(ssi_np);
+	if (!ssi_pdev) {
+		dev_err(dev, "failed to find SSI platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto fail;
+	}
+
+	data->dev = dev;
+
+	data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+
+	data->dai.name = "hifi";
+	data->dai.stream_name = "hifi";
+	data->dai.codec_dai_name = "es8328-hifi-analog";
+	data->dai.codec_of_node = codec_np;
+	data->dai.cpu_of_node = ssi_np;
+	data->dai.platform_of_node = ssi_np;
+	data->dai.init = &imx_es8328_dai_init;
+	data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			    SND_SOC_DAIFMT_CBM_CFM;
+
+	data->card.dev = dev;
+	data->card.dapm_widgets = imx_es8328_dapm_widgets;
+	data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets);
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret) {
+		dev_err(dev, "Unable to parse card name\n");
+		goto fail;
+	}
+	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+	if (ret) {
+		dev_err(dev, "Unable to parse routing: %d\n", ret);
+		goto fail;
+	}
+	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
+	data->card.dai_link = &data->dai;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(dev, "Unable to register: %d\n", ret);
+		goto fail;
+	}
+
+	platform_set_drvdata(pdev, data);
+fail:
+	of_node_put(ssi_np);
+	of_node_put(codec_np);
+
+	return ret;
+}
+
+static int imx_es8328_remove(struct platform_device *pdev)
+{
+	struct imx_es8328_data *data = platform_get_drvdata(pdev);
+
+	snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
+				headset_jack_gpios);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_es8328_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-es8328", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids);
+
+static struct platform_driver imx_es8328_driver = {
+	.driver = {
+		.name = "imx-es8328",
+		.of_match_table = imx_es8328_dt_ids,
+	},
+	.probe = imx_es8328_probe,
+	.remove = imx_es8328_remove,
+};
+module_platform_driver(imx_es8328_driver);
+
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-audio-es8328");