From 5ad8865b009bc8ad35adcbcb60c0679d437e8036 Mon Sep 17 00:00:00 2001
From: Jyri Sarha <jsarha@ti.com>
Date: Mon, 27 Jan 2014 17:37:51 +0200
Subject: ASoC: davinci-evm: Add named clock reference to DT bindings

The referenced clock is used to get codec clock rate and the clock is
disabled and enabled in startup and shutdown snd_soc_ops call
backs. The change is also documented in DT bindigs document.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
cc: bcousson@baylibre.com
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
index 865178d5cdf3..963e100514c2 100644
--- a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
@@ -5,12 +5,19 @@ Required properties:
 - ti,model : The user-visible name of this sound complex.
 - ti,audio-codec : The phandle of the TLV320AIC3x audio codec
 - ti,mcasp-controller : The phandle of the McASP controller
-- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec
 - ti,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 for sources and
   sinks are the codec's pins, and the jacks on the board:
 
+Optional properties:
+- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec.
+- clocks : Reference to the master clock
+- clock-names : The clock should be named "mclk"
+- Either codec-clock-rate or the codec-clock reference has to be defined. If
+  the both are defined the driver attempts to set referenced clock to the
+  defined rate and takes the rate from the clock reference.
+
   Board connectors:
 
   * Headphone Jack
-- 
cgit v1.2.3


From 4d16700dd926d4c4a66a91a138c34eef4fd342b4 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Mon, 27 Jan 2014 13:03:08 +0100
Subject: ASoC: tlv320aic32x4: DT support

Add DT support for this codec. The bindings differ a bit from the aic3x
codec bindings, so I created a new binding documentation.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/tlv320aic32x4.txt    | 18 ++++++++++++++++
 sound/soc/codecs/tlv320aic32x4.c                   | 25 ++++++++++++++++++++++
 2 files changed, 43 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
new file mode 100644
index 000000000000..db0551088cc4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -0,0 +1,18 @@
+Texas Instruments - tlv320aic32x4 Codec module
+
+The tlv320aic32x4 serial control bus communicates through I2C protocols
+
+Required properties:
+ - compatible: Should be "ti,tlv320aic32x4"
+ - reg: I2C slave address
+
+Optional properties:
+ - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
+
+
+Example:
+
+codec: tlv320aic32x4@18 {
+	compatible = "ti,tlv320aic32x4";
+	reg = <0x18>;
+};
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index c541213b1edf..1dd50e48934c 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -29,6 +29,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/gpio.h>
+#include <linux/of_gpio.h>
 #include <linux/i2c.h>
 #include <linux/cdev.h>
 #include <linux/slab.h>
@@ -669,11 +670,22 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
 };
 
+static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4,
+		struct device_node *np)
+{
+	aic32x4->swapdacs = false;
+	aic32x4->micpga_routing = 0;
+	aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0);
+
+	return 0;
+}
+
 static int aic32x4_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
 	struct aic32x4_pdata *pdata = i2c->dev.platform_data;
 	struct aic32x4_priv *aic32x4;
+	struct device_node *np = i2c->dev.of_node;
 	int ret;
 
 	aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv),
@@ -692,6 +704,12 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
 		aic32x4->swapdacs = pdata->swapdacs;
 		aic32x4->micpga_routing = pdata->micpga_routing;
 		aic32x4->rstn_gpio = pdata->rstn_gpio;
+	} else if (np) {
+		ret = aic32x4_parse_dt(aic32x4, np);
+		if (ret) {
+			dev_err(&i2c->dev, "Failed to parse DT node\n");
+			return ret;
+		}
 	} else {
 		aic32x4->power_cfg = 0;
 		aic32x4->swapdacs = false;
@@ -723,10 +741,17 @@ static const struct i2c_device_id aic32x4_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
 
+static const struct of_device_id aic32x4_of_id[] = {
+	{ .compatible = "ti,tlv320aic32x4", },
+	{ /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
 static struct i2c_driver aic32x4_i2c_driver = {
 	.driver = {
 		.name = "tlv320aic32x4",
 		.owner = THIS_MODULE,
+		.of_match_table = aic32x4_of_id,
 	},
 	.probe =    aic32x4_i2c_probe,
 	.remove =   aic32x4_i2c_remove,
-- 
cgit v1.2.3


From 5a3af1293194d07610fd7fdf680b3e7f916dca84 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 6 Feb 2014 12:03:27 +0000
Subject: ASoC: pcm512x: Add PCM512x driver

The PCM512x devices are a family of monolithic CMOS integrated circuits
that include a stereo digital-to-analog converter and additional support
circuitry.

This is an initial driver which supports some core functionality for the
device which covers common use cases but does not cover all features.
Currently only slave clocking modes with automatic clock configuration
are supported and most of the DSP configuration for the device is not
enabled.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/pcm512x.txt          |  30 +
 sound/soc/codecs/Kconfig                           |   4 +
 sound/soc/codecs/Makefile                          |   2 +
 sound/soc/codecs/pcm512x.c                         | 672 +++++++++++++++++++++
 sound/soc/codecs/pcm512x.h                         | 142 +++++
 5 files changed, 850 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt
 create mode 100644 sound/soc/codecs/pcm512x.c
 create mode 100644 sound/soc/codecs/pcm512x.h

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt
new file mode 100644
index 000000000000..faff75e64573
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm512x.txt
@@ -0,0 +1,30 @@
+PCM512x audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : One of "ti,pcm5121" or "ti,pcm5122"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+  - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the
+    device, as covered in bindings/regulator/regulator.txt
+
+Optional properties:
+
+  - clocks : A clock specifier for the clock connected as SCLK.  If this
+    is absent the device will be configured to clock from BCLK.
+
+Example:
+
+	pcm5122: pcm5122@4c {
+		compatible = "ti,pcm5122";
+		reg = <0x4c>;
+
+		AVDD-supply = <&reg_3v3_analog>;
+		DVDD-supply = <&reg_1v8>;
+		CPVDD-supply = <&reg_3v3>;
+	};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 983d087aa92a..56d0c2845680 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM1681 if I2C
 	select SND_SOC_PCM1792A if SPI_MASTER
 	select SND_SOC_PCM3008
+	select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_RT5640 if I2C
 	select SND_SOC_SGTL5000 if I2C
@@ -313,6 +314,9 @@ config SND_SOC_PCM1792A
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_PCM512x
+       tristate "Texas Instruments PCM512x CODECs"
+
 config SND_SOC_RT5631
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bc126764a44d..d3b536fc075d 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -46,6 +46,7 @@ snd-soc-hdmi-codec-objs := hdmi.o
 snd-soc-pcm1681-objs := pcm1681.o
 snd-soc-pcm1792a-codec-objs := pcm1792a.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm512x-objs := pcm512x.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-rt5640-objs := rt5640.o
 snd-soc-sgtl5000-objs := sgtl5000.o
@@ -179,6 +180,7 @@ obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
 obj-$(CONFIG_SND_SOC_PCM1792A)	+= snd-soc-pcm1792a-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM512x)	+= snd-soc-pcm512x.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_RT5640)	+= snd-soc-rt5640.o
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
new file mode 100644
index 000000000000..5ad3e9aa3cb4
--- /dev/null
+++ b/sound/soc/codecs/pcm512x.c
@@ -0,0 +1,672 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author:	Mark Brown <broonie@linaro.org>
+ *		Copyright 2014 Linaro Ltd
+ *
+ * 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.
+ *
+ * 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/init.h>
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "pcm512x.h"
+
+#define PCM512x_NUM_SUPPLIES 3
+static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = {
+	"AVDD",
+	"DVDD",
+	"CPVDD",
+};
+
+struct pcm512x_priv {
+	struct regmap *regmap;
+	struct clk *sclk;
+	struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES];
+	struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES];
+};
+
+/*
+ * We can't use the same notifier block for more than one supply and
+ * there's no way I can see to get from a callback to the caller
+ * except container_of().
+ */
+#define PCM512x_REGULATOR_EVENT(n) \
+static int pcm512x_regulator_event_##n(struct notifier_block *nb, \
+				      unsigned long event, void *data)    \
+{ \
+	struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \
+						    supply_nb[n]); \
+	if (event & REGULATOR_EVENT_DISABLE) { \
+		regcache_mark_dirty(pcm512x->regmap);	\
+		regcache_cache_only(pcm512x->regmap, true);	\
+	} \
+	return 0; \
+}
+
+PCM512x_REGULATOR_EVENT(0)
+PCM512x_REGULATOR_EVENT(1)
+PCM512x_REGULATOR_EVENT(2)
+
+static const struct reg_default pcm512x_reg_defaults[] = {
+	{ PCM512x_RESET,            0x00 },
+	{ PCM512x_POWER,            0x00 },
+	{ PCM512x_MUTE,             0x00 },
+	{ PCM512x_DSP,              0x00 },
+	{ PCM512x_PLL_REF,          0x00 },
+	{ PCM512x_DAC_ROUTING,      0x11 },
+	{ PCM512x_DSP_PROGRAM,      0x01 },
+	{ PCM512x_CLKDET,           0x00 },
+	{ PCM512x_AUTO_MUTE,        0x00 },
+	{ PCM512x_ERROR_DETECT,     0x00 },
+	{ PCM512x_DIGITAL_VOLUME_1, 0x00 },
+	{ PCM512x_DIGITAL_VOLUME_2, 0x30 },
+	{ PCM512x_DIGITAL_VOLUME_3, 0x30 },
+	{ PCM512x_DIGITAL_MUTE_1,   0x22 },
+	{ PCM512x_DIGITAL_MUTE_2,   0x00 },
+	{ PCM512x_DIGITAL_MUTE_3,   0x07 },
+};
+
+static bool pcm512x_readable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case PCM512x_RESET:
+	case PCM512x_POWER:
+	case PCM512x_MUTE:
+	case PCM512x_PLL_EN:
+	case PCM512x_SPI_MISO_FUNCTION:
+	case PCM512x_DSP:
+	case PCM512x_GPIO_EN:
+	case PCM512x_BCLK_LRCLK_CFG:
+	case PCM512x_DSP_GPIO_INPUT:
+	case PCM512x_MASTER_MODE:
+	case PCM512x_PLL_REF:
+	case PCM512x_PLL_COEFF_0:
+	case PCM512x_PLL_COEFF_1:
+	case PCM512x_PLL_COEFF_2:
+	case PCM512x_PLL_COEFF_3:
+	case PCM512x_PLL_COEFF_4:
+	case PCM512x_DSP_CLKDIV:
+	case PCM512x_DAC_CLKDIV:
+	case PCM512x_NCP_CLKDIV:
+	case PCM512x_OSR_CLKDIV:
+	case PCM512x_MASTER_CLKDIV_1:
+	case PCM512x_MASTER_CLKDIV_2:
+	case PCM512x_FS_SPEED_MODE:
+	case PCM512x_IDAC_1:
+	case PCM512x_IDAC_2:
+	case PCM512x_ERROR_DETECT:
+	case PCM512x_I2S_1:
+	case PCM512x_I2S_2:
+	case PCM512x_DAC_ROUTING:
+	case PCM512x_DSP_PROGRAM:
+	case PCM512x_CLKDET:
+	case PCM512x_AUTO_MUTE:
+	case PCM512x_DIGITAL_VOLUME_1:
+	case PCM512x_DIGITAL_VOLUME_2:
+	case PCM512x_DIGITAL_VOLUME_3:
+	case PCM512x_DIGITAL_MUTE_1:
+	case PCM512x_DIGITAL_MUTE_2:
+	case PCM512x_DIGITAL_MUTE_3:
+	case PCM512x_GPIO_OUTPUT_1:
+	case PCM512x_GPIO_OUTPUT_2:
+	case PCM512x_GPIO_OUTPUT_3:
+	case PCM512x_GPIO_OUTPUT_4:
+	case PCM512x_GPIO_OUTPUT_5:
+	case PCM512x_GPIO_OUTPUT_6:
+	case PCM512x_GPIO_CONTROL_1:
+	case PCM512x_GPIO_CONTROL_2:
+	case PCM512x_OVERFLOW:
+	case PCM512x_RATE_DET_1:
+	case PCM512x_RATE_DET_2:
+	case PCM512x_RATE_DET_3:
+	case PCM512x_RATE_DET_4:
+	case PCM512x_ANALOG_MUTE_DET:
+	case PCM512x_GPIN:
+	case PCM512x_DIGITAL_MUTE_DET:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool pcm512x_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case PCM512x_PLL_EN:
+	case PCM512x_OVERFLOW:
+	case PCM512x_RATE_DET_1:
+	case PCM512x_RATE_DET_2:
+	case PCM512x_RATE_DET_3:
+	case PCM512x_RATE_DET_4:
+	case PCM512x_ANALOG_MUTE_DET:
+	case PCM512x_GPIN:
+	case PCM512x_DIGITAL_MUTE_DET:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1);
+
+static const char *pcm512x_dsp_program_texts[] = {
+	"FIR interpolation with de-emphasis",
+	"Low latency IIR with de-emphasis",
+	"Fixed process flow",
+	"High attenuation with de-emphasis",
+	"Ringing-less low latency FIR",
+};
+
+static const unsigned int pcm512x_dsp_program_values[] = {
+	1,
+	2,
+	3,
+	5,
+	7,
+};
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program,
+					PCM512x_DSP_PROGRAM, 0, 0x1f,
+					pcm512x_dsp_program_texts,
+					pcm512x_dsp_program_values);
+
+static const char *pcm512x_clk_missing_text[] = {
+	"1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s"
+};
+
+static const struct soc_enum pcm512x_clk_missing =
+	SOC_ENUM_SINGLE(PCM512x_CLKDET, 0,  8, pcm512x_clk_missing_text);
+
+static const char *pcm512x_autom_text[] = {
+	"21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s"
+};
+
+static const struct soc_enum pcm512x_autom_l =
+	SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 8,
+			pcm512x_autom_text);
+
+static const struct soc_enum pcm512x_autom_r =
+	SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8,
+			pcm512x_autom_text);
+
+static const char *pcm512x_ramp_rate_text[] = {
+	"1 sample/update", "2 samples/update", "4 samples/update",
+	"Immediate"
+};
+
+static const struct soc_enum pcm512x_vndf =
+	SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4,
+			pcm512x_ramp_rate_text);
+
+static const struct soc_enum pcm512x_vnuf =
+	SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4,
+			pcm512x_ramp_rate_text);
+
+static const struct soc_enum pcm512x_vedf =
+	SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4,
+			pcm512x_ramp_rate_text);
+
+static const char *pcm512x_ramp_step_text[] = {
+	"4dB/step", "2dB/step", "1dB/step", "0.5dB/step"
+};
+
+static const struct soc_enum pcm512x_vnds =
+	SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4,
+			pcm512x_ramp_step_text);
+
+static const struct soc_enum pcm512x_vnus =
+	SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4,
+			pcm512x_ramp_step_text);
+
+static const struct soc_enum pcm512x_veds =
+	SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4,
+			pcm512x_ramp_step_text);
+
+static const struct snd_kcontrol_new pcm512x_controls[] = {
+SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+		 PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
+SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+	   PCM512x_RQMR_SHIFT, 1, 1),
+
+SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
+SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program),
+
+SOC_ENUM("Clock Missing Period", pcm512x_clk_missing),
+SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l),
+SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r),
+SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3,
+	   PCM512x_ACTL_SHIFT, 1, 0),
+SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT,
+	   PCM512x_AMLR_SHIFT, 1, 0),
+
+SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf),
+SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds),
+SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf),
+SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus),
+SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf),
+SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds),
+};
+
+static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_OUTPUT("OUTL"),
+SND_SOC_DAPM_OUTPUT("OUTR"),
+};
+
+static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = {
+	{ "DACL", NULL, "Playback" },
+	{ "DACR", NULL, "Playback" },
+
+	{ "OUTL", NULL, "DACL" },
+	{ "OUTR", NULL, "DACR" },
+};
+
+static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev);
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+					 PCM512x_RQST, 0);
+		if (ret != 0) {
+			dev_err(codec->dev, "Failed to remove standby: %d\n",
+				ret);
+			return ret;
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+					 PCM512x_RQST, PCM512x_RQST);
+		if (ret != 0) {
+			dev_err(codec->dev, "Failed to request standby: %d\n",
+				ret);
+			return ret;
+		}
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver pcm512x_dai = {
+	.name = "pcm512x-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			   SNDRV_PCM_FMTBIT_S24_LE |
+			   SNDRV_PCM_FMTBIT_S32_LE
+	},
+};
+
+static struct snd_soc_codec_driver pcm512x_codec_driver = {
+	.set_bias_level = pcm512x_set_bias_level,
+	.idle_bias_off = true,
+
+	.controls = pcm512x_controls,
+	.num_controls = ARRAY_SIZE(pcm512x_controls),
+	.dapm_widgets = pcm512x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets),
+	.dapm_routes = pcm512x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes),
+};
+
+static const struct regmap_config pcm512x_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.readable_reg = pcm512x_readable,
+	.volatile_reg = pcm512x_volatile,
+
+	.max_register = PCM512x_MAX_REGISTER,
+	.reg_defaults = pcm512x_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static const struct of_device_id pcm512x_of_match[] = {
+	{ .compatible = "ti,pcm5121", },
+	{ .compatible = "ti,pcm5122", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, pcm512x_of_match);
+
+static int pcm512x_probe(struct device *dev, struct regmap *regmap)
+{
+	struct pcm512x_priv *pcm512x;
+	int i, ret;
+
+	pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL);
+	if (!pcm512x)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, pcm512x);
+	pcm512x->regmap = regmap;
+
+	for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++)
+		pcm512x->supplies[i].supply = pcm512x_supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies),
+				      pcm512x->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to get supplies: %d\n", ret);
+		return ret;
+	}
+
+	pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0;
+	pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1;
+	pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2;
+
+	for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) {
+		ret = regulator_register_notifier(pcm512x->supplies[i].consumer,
+						  &pcm512x->supply_nb[i]);
+		if (ret != 0) {
+			dev_err(dev,
+				"Failed to register regulator notifier: %d\n",
+				ret);
+		}
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies),
+				    pcm512x->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	/* Reset the device, verifying I/O in the process for I2C */
+	ret = regmap_write(regmap, PCM512x_RESET,
+			   PCM512x_RSTM | PCM512x_RSTR);
+	if (ret != 0) {
+		dev_err(dev, "Failed to reset device: %d\n", ret);
+		goto err;
+	}
+
+	ret = regmap_write(regmap, PCM512x_RESET, 0);
+	if (ret != 0) {
+		dev_err(dev, "Failed to reset device: %d\n", ret);
+		goto err;
+	}
+
+	pcm512x->sclk = devm_clk_get(dev, NULL);
+	if (IS_ERR(pcm512x->sclk)) {
+		if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
+			return -EPROBE_DEFER;
+
+		dev_info(dev, "No SCLK, using BCLK: %ld\n",
+			 PTR_ERR(pcm512x->sclk));
+
+		/* Disable reporting of missing SCLK as an error */
+		regmap_update_bits(regmap, PCM512x_ERROR_DETECT,
+				   PCM512x_IDCH, PCM512x_IDCH);
+
+		/* Switch PLL input to BCLK */
+		regmap_update_bits(regmap, PCM512x_PLL_REF,
+				   PCM512x_SREF, PCM512x_SREF);
+	} else {
+		ret = clk_prepare_enable(pcm512x->sclk);
+		if (ret != 0) {
+			dev_err(dev, "Failed to enable SCLK: %d\n", ret);
+			return ret;
+		}
+	}
+
+	/* Default to standby mode */
+	ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+				 PCM512x_RQST, PCM512x_RQST);
+	if (ret != 0) {
+		dev_err(dev, "Failed to request standby: %d\n",
+			ret);
+		goto err_clk;
+	}
+
+	pm_runtime_set_active(dev);
+	pm_runtime_enable(dev);
+	pm_runtime_idle(dev);
+
+	ret = snd_soc_register_codec(dev, &pcm512x_codec_driver,
+				    &pcm512x_dai, 1);
+	if (ret != 0) {
+		dev_err(dev, "Failed to register CODEC: %d\n", ret);
+		goto err_pm;
+	}
+
+	return 0;
+
+err_pm:
+	pm_runtime_disable(dev);
+err_clk:
+	if (!IS_ERR(pcm512x->sclk))
+		clk_disable_unprepare(pcm512x->sclk);
+err:
+	regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+				     pcm512x->supplies);
+	return ret;
+}
+
+static void pcm512x_remove(struct device *dev)
+{
+	struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+
+	snd_soc_unregister_codec(dev);
+	pm_runtime_disable(dev);
+	if (!IS_ERR(pcm512x->sclk))
+		clk_disable_unprepare(pcm512x->sclk);
+	regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+			       pcm512x->supplies);
+}
+
+static int pcm512x_suspend(struct device *dev)
+{
+	struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+	int ret;
+
+	ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+				 PCM512x_RQPD, PCM512x_RQPD);
+	if (ret != 0) {
+		dev_err(dev, "Failed to request power down: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+				     pcm512x->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to disable supplies: %d\n", ret);
+		return ret;
+	}
+
+	if (!IS_ERR(pcm512x->sclk))
+		clk_disable_unprepare(pcm512x->sclk);
+
+	return 0;
+}
+
+static int pcm512x_resume(struct device *dev)
+{
+	struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+	int ret;
+
+	if (!IS_ERR(pcm512x->sclk)) {
+		ret = clk_prepare_enable(pcm512x->sclk);
+		if (ret != 0) {
+			dev_err(dev, "Failed to enable SCLK: %d\n", ret);
+			return ret;
+		}
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies),
+				    pcm512x->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(pcm512x->regmap, false);
+	ret = regcache_sync(pcm512x->regmap);
+	if (ret != 0) {
+		dev_err(dev, "Failed to sync cache: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+				 PCM512x_RQPD, 0);
+	if (ret != 0) {
+		dev_err(dev, "Failed to remove power down: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct dev_pm_ops pcm512x_pm_ops = {
+	SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL)
+};
+
+#if IS_ENABLED(CONFIG_I2C)
+static int pcm512x_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct regmap *regmap;
+
+	regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	return pcm512x_probe(&i2c->dev, regmap);
+}
+
+static int pcm512x_i2c_remove(struct i2c_client *i2c)
+{
+	pcm512x_remove(&i2c->dev);
+	return 0;
+}
+
+static const struct i2c_device_id pcm512x_i2c_id[] = {
+	{ "pcm5121", },
+	{ "pcm5122", },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id);
+
+static struct i2c_driver pcm512x_i2c_driver = {
+	.probe 		= pcm512x_i2c_probe,
+	.remove 	= pcm512x_i2c_remove,
+	.id_table	= pcm512x_i2c_id,
+	.driver		= {
+		.name	= "pcm512x",
+		.owner	= THIS_MODULE,
+		.of_match_table = pcm512x_of_match,
+		.pm     = &pcm512x_pm_ops,
+	},
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int pcm512x_spi_probe(struct spi_device *spi)
+{
+	struct regmap *regmap;
+	int ret;
+
+	regmap = devm_regmap_init_spi(spi, &pcm512x_regmap);
+	if (IS_ERR(regmap)) {
+		ret = PTR_ERR(regmap);
+		return ret;
+	}
+
+	return pcm512x_probe(&spi->dev, regmap);
+}
+
+static int pcm512x_spi_remove(struct spi_device *spi)
+{
+	pcm512x_remove(&spi->dev);
+	return 0;
+}
+
+static const struct spi_device_id pcm512x_spi_id[] = {
+	{ "pcm5121", },
+	{ "pcm5122", },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, pcm512x_spi_id);
+
+static struct spi_driver pcm512x_spi_driver = {
+	.probe		= pcm512x_spi_probe,
+	.remove		= pcm512x_spi_remove,
+	.id_table	= pcm512x_spi_id,
+	.driver = {
+		.name	= "pcm512x",
+		.owner	= THIS_MODULE,
+		.of_match_table = pcm512x_of_match,
+		.pm     = &pcm512x_pm_ops,
+	},
+};
+#endif
+
+static int __init pcm512x_modinit(void)
+{
+	int ret = 0;
+
+#if IS_ENABLED(CONFIG_I2C)
+	ret = i2c_add_driver(&pcm512x_i2c_driver);
+	if (ret) {
+		printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n",
+		       ret);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&pcm512x_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n",
+		       ret);
+	}
+#endif
+	return ret;
+}
+module_init(pcm512x_modinit);
+
+static void __exit pcm512x_exit(void)
+{
+#if IS_ENABLED(CONFIG_I2C)
+	i2c_del_driver(&pcm512x_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&pcm512x_spi_driver);
+#endif
+}
+module_exit(pcm512x_exit);
+
+MODULE_DESCRIPTION("ASoC PCM512x codec driver");
+MODULE_AUTHOR("Mark Brown <broonie@linaro.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h
new file mode 100644
index 000000000000..b2f518ecb35c
--- /dev/null
+++ b/sound/soc/codecs/pcm512x.h
@@ -0,0 +1,142 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author:	Mark Brown <broonie@linaro.org>
+ *		Copyright 2014 Linaro Ltd
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef _SND_SOC_PCM512X
+#define _SND_SOC_PCM512X
+
+#define PCM512x_PAGE_0_BASE 0
+
+#define PCM512x_PAGE              0
+
+#define PCM512x_RESET             (PCM512x_PAGE_0_BASE +   1)
+#define PCM512x_POWER             (PCM512x_PAGE_0_BASE +   2)
+#define PCM512x_MUTE              (PCM512x_PAGE_0_BASE +   3)
+#define PCM512x_PLL_EN            (PCM512x_PAGE_0_BASE +   4)
+#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE +   6)
+#define PCM512x_DSP               (PCM512x_PAGE_0_BASE +   7)
+#define PCM512x_GPIO_EN           (PCM512x_PAGE_0_BASE +   8)
+#define PCM512x_BCLK_LRCLK_CFG    (PCM512x_PAGE_0_BASE +   9)
+#define PCM512x_DSP_GPIO_INPUT    (PCM512x_PAGE_0_BASE +  10)
+#define PCM512x_MASTER_MODE       (PCM512x_PAGE_0_BASE +  12)
+#define PCM512x_PLL_REF           (PCM512x_PAGE_0_BASE +  13)
+#define PCM512x_PLL_COEFF_0       (PCM512x_PAGE_0_BASE +  20)
+#define PCM512x_PLL_COEFF_1       (PCM512x_PAGE_0_BASE +  21)
+#define PCM512x_PLL_COEFF_2       (PCM512x_PAGE_0_BASE +  22)
+#define PCM512x_PLL_COEFF_3       (PCM512x_PAGE_0_BASE +  23)
+#define PCM512x_PLL_COEFF_4       (PCM512x_PAGE_0_BASE +  24)
+#define PCM512x_DSP_CLKDIV        (PCM512x_PAGE_0_BASE +  27)
+#define PCM512x_DAC_CLKDIV        (PCM512x_PAGE_0_BASE +  28)
+#define PCM512x_NCP_CLKDIV        (PCM512x_PAGE_0_BASE +  29)
+#define PCM512x_OSR_CLKDIV        (PCM512x_PAGE_0_BASE +  30)
+#define PCM512x_MASTER_CLKDIV_1   (PCM512x_PAGE_0_BASE +  32)
+#define PCM512x_MASTER_CLKDIV_2   (PCM512x_PAGE_0_BASE +  33)
+#define PCM512x_FS_SPEED_MODE     (PCM512x_PAGE_0_BASE +  34)
+#define PCM512x_IDAC_1            (PCM512x_PAGE_0_BASE +  35)
+#define PCM512x_IDAC_2            (PCM512x_PAGE_0_BASE +  36)
+#define PCM512x_ERROR_DETECT      (PCM512x_PAGE_0_BASE +  37)
+#define PCM512x_I2S_1             (PCM512x_PAGE_0_BASE +  40)
+#define PCM512x_I2S_2             (PCM512x_PAGE_0_BASE +  41)
+#define PCM512x_DAC_ROUTING       (PCM512x_PAGE_0_BASE +  42)
+#define PCM512x_DSP_PROGRAM       (PCM512x_PAGE_0_BASE +  43)
+#define PCM512x_CLKDET            (PCM512x_PAGE_0_BASE +  44)
+#define PCM512x_AUTO_MUTE         (PCM512x_PAGE_0_BASE +  59)
+#define PCM512x_DIGITAL_VOLUME_1  (PCM512x_PAGE_0_BASE +  60)
+#define PCM512x_DIGITAL_VOLUME_2  (PCM512x_PAGE_0_BASE +  61)
+#define PCM512x_DIGITAL_VOLUME_3  (PCM512x_PAGE_0_BASE +  62)
+#define PCM512x_DIGITAL_MUTE_1    (PCM512x_PAGE_0_BASE +  63)
+#define PCM512x_DIGITAL_MUTE_2    (PCM512x_PAGE_0_BASE +  64)
+#define PCM512x_DIGITAL_MUTE_3    (PCM512x_PAGE_0_BASE +  65)
+#define PCM512x_GPIO_OUTPUT_1     (PCM512x_PAGE_0_BASE +  80)
+#define PCM512x_GPIO_OUTPUT_2     (PCM512x_PAGE_0_BASE +  81)
+#define PCM512x_GPIO_OUTPUT_3     (PCM512x_PAGE_0_BASE +  82)
+#define PCM512x_GPIO_OUTPUT_4     (PCM512x_PAGE_0_BASE +  83)
+#define PCM512x_GPIO_OUTPUT_5     (PCM512x_PAGE_0_BASE +  84)
+#define PCM512x_GPIO_OUTPUT_6     (PCM512x_PAGE_0_BASE +  85)
+#define PCM512x_GPIO_CONTROL_1    (PCM512x_PAGE_0_BASE +  86)
+#define PCM512x_GPIO_CONTROL_2    (PCM512x_PAGE_0_BASE +  87)
+#define PCM512x_OVERFLOW          (PCM512x_PAGE_0_BASE +  90)
+#define PCM512x_RATE_DET_1        (PCM512x_PAGE_0_BASE +  91)
+#define PCM512x_RATE_DET_2        (PCM512x_PAGE_0_BASE +  92)
+#define PCM512x_RATE_DET_3        (PCM512x_PAGE_0_BASE +  93)
+#define PCM512x_RATE_DET_4        (PCM512x_PAGE_0_BASE +  94)
+#define PCM512x_ANALOG_MUTE_DET   (PCM512x_PAGE_0_BASE + 108)
+#define PCM512x_GPIN              (PCM512x_PAGE_0_BASE + 119)
+#define PCM512x_DIGITAL_MUTE_DET  (PCM512x_PAGE_0_BASE + 120)
+
+#define PCM512x_MAX_REGISTER      (PCM512x_PAGE_0_BASE + 120)
+
+/* Page 0, Register 1 - reset */
+#define PCM512x_RSTR (1 << 0)
+#define PCM512x_RSTM (1 << 4)
+
+/* Page 0, Register 2 - power */
+#define PCM512x_RQPD       (1 << 0)
+#define PCM512x_RQPD_SHIFT 0
+#define PCM512x_RQST       (1 << 4)
+#define PCM512x_RQST_SHIFT 4
+
+/* Page 0, Register 3 - mute */
+#define PCM512x_RQMR_SHIFT 0
+#define PCM512x_RQML_SHIFT 4
+
+/* Page 0, Register 4 - PLL */
+#define PCM512x_PLCE       (1 << 0)
+#define PCM512x_RLCE_SHIFT 0
+#define PCM512x_PLCK       (1 << 4)
+#define PCM512x_PLCK_SHIFT 4
+
+/* Page 0, Register 7 - DSP */
+#define PCM512x_SDSL       (1 << 0)
+#define PCM512x_SDSL_SHIFT 0
+#define PCM512x_DEMP       (1 << 4)
+#define PCM512x_DEMP_SHIFT 4
+
+/* Page 0, Register 13 - PLL reference */
+#define PCM512x_SREF (1 << 4)
+
+/* Page 0, Register 37 - Error detection */
+#define PCM512x_IPLK (1 << 0)
+#define PCM512x_DCAS (1 << 1)
+#define PCM512x_IDCM (1 << 2)
+#define PCM512x_IDCH (1 << 3)
+#define PCM512x_IDSK (1 << 4)
+#define PCM512x_IDBK (1 << 5)
+#define PCM512x_IDFS (1 << 6)
+
+/* Page 0, Register 42 - DAC routing */
+#define PCM512x_AUPR_SHIFT 0
+#define PCM512x_AUPL_SHIFT 4
+
+/* Page 0, Register 59 - auto mute */
+#define PCM512x_ATMR_SHIFT 0
+#define PCM512x_ATML_SHIFT 4
+
+/* Page 0, Register 63 - ramp rates */
+#define PCM512x_VNDF_SHIFT 6
+#define PCM512x_VNDS_SHIFT 4
+#define PCM512x_VNUF_SHIFT 2
+#define PCM512x_VNUS_SHIFT 0
+
+/* Page 0, Register 64 - emergency ramp rates */
+#define PCM512x_VEDF_SHIFT 6
+#define PCM512x_VEDS_SHIFT 4
+
+/* Page 0, Register 65 - Digital mute enables */
+#define PCM512x_ACTL_SHIFT 2
+#define PCM512x_AMLE_SHIFT 1
+#define PCM512x_AMLR_SHIFT 0
+
+#endif
-- 
cgit v1.2.3


From ac7f4820301b6b6029f01e4c276c2cd42272d9cb Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Sat, 8 Feb 2014 15:59:54 +0800
Subject: ASoC: binding: add widgets.txt

Suggested-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/widgets.txt | 20 ++++++++++++++++++++
 1 file changed, 20 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/widgets.txt b/Documentation/devicetree/bindings/sound/widgets.txt
new file mode 100644
index 000000000000..b6de5ba3b2de
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/widgets.txt
@@ -0,0 +1,20 @@
+Widgets:
+
+This mainly specifies audio off-codec DAPM widgets.
+
+Each entry is a pair of strings in DT:
+
+	"template-wname", "user-supplied-wname"
+
+The "template-wname" being the template widget name and currently includes:
+"Microphone", "Line", "Headphone" and "Speaker".
+
+The "user-supplied-wname" being the user specified widget name.
+
+For instance:
+	simple-audio-widgets =
+		"Microphone", "Microphone Jack",
+		"Line", "Line In Jack",
+		"Line", "Line Out Jack",
+		"Headphone", "Headphone Jack",
+		"Speaker", "Speaker External";
-- 
cgit v1.2.3


From 66841345073b049c0c194486bac4d7f07266a57e Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Sat, 8 Feb 2014 15:59:55 +0800
Subject: ASoC: simple-card: for new properties documenting and usage

This add the following three new properties documenting and usage
for simple card:

"simple-audio-card,name",
"simple-audio-card,widgets",

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/simple-card.txt | 12 ++++++++++--
 1 file changed, 10 insertions(+), 2 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index 19c84df5fffa..05273583490c 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -8,9 +8,12 @@ Required properties:
 
 Optional properties:
 
+- simple-audio-card,name		: User specified audio sound card name, one string
+					  property.
 - simple-audio-card,format		: CPU/CODEC common audio format.
 					  "i2s", "right_j", "left_j" , "dsp_a"
 					  "dsp_b", "ac97", "pdm", "msb", "lsb"
+- simple-audio-card,widgets		: Please refer to widgets.txt.
 - simple-audio-card,routing		: A list of the connections between audio components.
 					  Each entry is a pair of strings, the first being the
 					  connection's sink, the second being the connection's
@@ -42,11 +45,16 @@ Example:
 
 sound {
 	compatible = "simple-audio-card";
+	simple-audio-card,name = "VF610-Tower-Sound-Card";
 	simple-audio-card,format = "left_j";
+	simple-audio-card,widgets =
+		"Microphone", "Microphone Jack",
+		"Headphone", "Headphone Jack",
+		"Speaker", "External Speaker";
 	simple-audio-card,routing =
-		"MIC_IN", "Mic Jack",
+		"MIC_IN", "Microphone Jack",
 		"Headphone Jack", "HP_OUT",
-		"Ext Spk", "LINE_OUT";
+		"External Speaker", "LINE_OUT";
 
 	simple-audio-card,cpu {
 		sound-dai = <&sh_fsi2 0>;
-- 
cgit v1.2.3


From 9f10b36ffde2b732def037c1e764a0c71745a372 Mon Sep 17 00:00:00 2001
From: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Date: Thu, 6 Feb 2014 18:03:09 +0000
Subject: ASoC: da9055: Add DT support for CODEC

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/da9055.txt | 22 ++++++++++++++++++++++
 sound/soc/codecs/da9055.c                          |  8 ++++++++
 2 files changed, 30 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/da9055.txt b/Documentation/devicetree/bindings/sound/da9055.txt
new file mode 100644
index 000000000000..ed1b7cc6f249
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da9055.txt
@@ -0,0 +1,22 @@
+* Dialog DA9055 Audio CODEC
+
+DA9055 provides Audio CODEC support (I2C only).
+
+The Audio CODEC device in DA9055 has it's own I2C address which is configurable,
+so the device is instantiated separately from the PMIC (MFD) device.
+
+For details on accompanying PMIC I2C device, see the following:
+Documentation/devicetree/bindings/mfd/da9055.txt
+
+Required properties:
+
+  - compatible: "dlg,da9055-codec"
+  - reg: Specifies the I2C slave address
+
+
+Example:
+
+	codec: da9055-codec@1a {
+		compatible = "dlg,da9055-codec";
+		reg = <0x1a>;
+	};
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 422812613a28..be31f3cfd46e 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -18,6 +18,8 @@
 #include <linux/regmap.h>
 #include <linux/slab.h>
 #include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -1536,11 +1538,17 @@ static const struct i2c_device_id da9055_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
 
+static const struct of_device_id da9055_of_match[] = {
+	{ .compatible = "dlg,da9055-codec", },
+	{ }
+};
+
 /* I2C codec control layer */
 static struct i2c_driver da9055_i2c_driver = {
 	.driver = {
 		.name = "da9055-codec",
 		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(da9055_of_match),
 	},
 	.probe		= da9055_i2c_probe,
 	.remove		= da9055_remove,
-- 
cgit v1.2.3


From dfd72a68aa0f6cf87575f3181319bde8a2d4c01b Mon Sep 17 00:00:00 2001
From: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Date: Thu, 30 Jan 2014 18:14:05 +0100
Subject: ASoC: cs42l51: add Device Tree binding to cs42l51

This commit adds a trivial Device Tree binding to the I2C-based
cs42l51 sound codec, so that it can be used from Device Tree based
platforms.

Signed-off-by: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/i2c/trivial-devices.txt | 1 +
 sound/soc/codecs/cs42l51.c                                | 7 +++++++
 2 files changed, 8 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/i2c/trivial-devices.txt b/Documentation/devicetree/bindings/i2c/trivial-devices.txt
index 1a1ac2e560e9..f47e56bcf78d 100644
--- a/Documentation/devicetree/bindings/i2c/trivial-devices.txt
+++ b/Documentation/devicetree/bindings/i2c/trivial-devices.txt
@@ -18,6 +18,7 @@ atmel,24c02		i2c serial eeprom  (24cxx)
 atmel,at97sc3204t	i2c trusted platform module (TPM)
 capella,cm32181		CM32181: Ambient Light Sensor
 catalyst,24c32		i2c serial eeprom
+cirrus,cs42l51		Cirrus Logic CS42L51 audio codec
 dallas,ds1307		64 x 8, Serial, I2C Real-Time Clock
 dallas,ds1338		I2C RTC with 56-Byte NV RAM
 dallas,ds1339		I2C Serial Real-Time Clock
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 6e9ea8379a91..824cdf4d4974 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -600,10 +600,17 @@ static const struct i2c_device_id cs42l51_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, cs42l51_id);
 
+static const struct of_device_id cs42l51_of_match[] = {
+	{ .compatible = "cirrus,cs42l51", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, cs42l51_of_match);
+
 static struct i2c_driver cs42l51_i2c_driver = {
 	.driver = {
 		.name = "cs42l51-codec",
 		.owner = THIS_MODULE,
+		.of_match_table = cs42l51_of_match,
 	},
 	.id_table = cs42l51_id,
 	.probe = cs42l51_i2c_probe,
-- 
cgit v1.2.3


From 9a0d5113ac0ee513224ca2e5011b3a566de16207 Mon Sep 17 00:00:00 2001
From: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Date: Thu, 30 Jan 2014 18:14:06 +0100
Subject: ASoC: kirkwood: enable Kirkwood driver for mvebu platforms

The audio unit found in the Armada 370 SoC is similar to the one used
in the Marvell Kirkwood and Marvell Dove SoCs. Therefore, this commit
allows the Kirkwood audio driver to be built on mvebu platforms, and
adds an additional compatible string to identify the Armada 370
variant of the audio unit.

Signed-off-by: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/mvebu-audio.txt | 1 +
 sound/soc/kirkwood/Kconfig                              | 2 +-
 sound/soc/kirkwood/kirkwood-i2s.c                       | 1 +
 3 files changed, 3 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
index f0062c5871b4..cb8c07c81ce4 100644
--- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@@ -5,6 +5,7 @@ Required properties:
 - compatible:
   "marvell,kirkwood-audio" for Kirkwood platforms
   "marvell,dove-audio" for Dove platforms
+  "marvell,armada370-audio" for Armada 370 platforms
 
 - reg: physical base address of the controller and length of memory mapped
   region.
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 78ed4a42ad21..764a0ef6b268 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,6 @@
 config SND_KIRKWOOD_SOC
 	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
-	depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
+	depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3920a5e8125f..9f842222e798 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -633,6 +633,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
 static struct of_device_id mvebu_audio_of_match[] = {
 	{ .compatible = "marvell,kirkwood-audio" },
 	{ .compatible = "marvell,dove-audio" },
+	{ .compatible = "marvell,armada370-audio" },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
-- 
cgit v1.2.3


From 393aa9c1cc514774332d7bc861307a76206e358d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Jan 2014 12:51:12 +0100
Subject: ALSA: Mandate to pass a device pointer at card creation time

This is a part of preliminary works for modernizing the ALSA device
structure.  So far, we set card->dev at later point after the object
creation.  Because of this, the core layer doesn't always know which
device is being handled before it's actually registered, and it makes
impossible to show the device in error messages, for example.  The
first goal is to achieve a proper struct device initialization at the
very beginning of probing.

As a first step, this patch introduces snd_card_new() function (yes
there was the same named function in the very past), in order to
receive the parent device pointer from the very beginning.
snd_card_create() is marked as deprecated.

At this point, there is no functional change other than that.  The
actual change of the device creation scheme will follow later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/DocBook/writing-an-alsa-driver.tmpl | 72 +++++++++--------------
 include/sound/core.h                              | 13 +++-
 sound/core/init.c                                 |  8 ++-
 3 files changed, 42 insertions(+), 51 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 06741e925985..d0056a4e9c53 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -468,8 +468,6 @@
                   return err;
           }
 
-          snd_card_set_dev(card, &pci->dev);
-
           *rchip = chip;
           return 0;
   }
@@ -492,7 +490,8 @@
           }
 
           /* (2) */
-          err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+          err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+                             0, &card);
           if (err < 0)
                   return err;
 
@@ -591,7 +590,8 @@
   struct snd_card *card;
   int err;
   ....
-  err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+  err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+                     0, &card);
 ]]>
             </programlisting>
           </informalexample>
@@ -809,28 +809,34 @@
 
       <para>
         As mentioned above, to create a card instance, call
-      <function>snd_card_create()</function>.
+      <function>snd_card_new()</function>.
 
         <informalexample>
           <programlisting>
 <![CDATA[
   struct snd_card *card;
   int err;
-  err = snd_card_create(index, id, module, extra_size, &card);
+  err = snd_card_new(&pci->dev, index, id, module, extra_size, &card);
 ]]>
           </programlisting>
         </informalexample>
       </para>
 
       <para>
-        The function takes five arguments, the card-index number, the
-        id string, the module pointer (usually
+        The function takes six arguments: the parent device pointer,
+        the card-index number, the id string, the module pointer (usually
         <constant>THIS_MODULE</constant>),
         the size of extra-data space, and the pointer to return the
         card instance.  The extra_size argument is used to
         allocate card-&gt;private_data for the
         chip-specific data.  Note that these data
-        are allocated by <function>snd_card_create()</function>.
+        are allocated by <function>snd_card_new()</function>.
+      </para>
+
+      <para>
+	The first argument, the pointer of struct
+	<structname>device</structname>, specifies the parent device.
+	For PCI devices, typically &amp;pci-&gt; is passed there.
       </para>
     </section>
 
@@ -916,16 +922,16 @@
       </para>
 
       <section id="card-management-chip-specific-snd-card-new">
-        <title>1. Allocating via <function>snd_card_create()</function>.</title>
+        <title>1. Allocating via <function>snd_card_new()</function>.</title>
         <para>
           As mentioned above, you can pass the extra-data-length
-	  to the 4th argument of <function>snd_card_create()</function>, i.e.
+	  to the 5th argument of <function>snd_card_new()</function>, i.e.
 
           <informalexample>
             <programlisting>
 <![CDATA[
-  err = snd_card_create(index[dev], id[dev], THIS_MODULE,
-                        sizeof(struct mychip), &card);
+  err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+                     sizeof(struct mychip), &card);
 ]]>
             </programlisting>
           </informalexample>
@@ -954,7 +960,7 @@
 
         <para>
           After allocating a card instance via
-          <function>snd_card_create()</function> (with
+          <function>snd_card_new()</function> (with
           <constant>0</constant> on the 4th arg), call
           <function>kzalloc()</function>. 
 
@@ -963,7 +969,8 @@
 <![CDATA[
   struct snd_card *card;
   struct mychip *chip;
-  err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+  err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+                     0, &card);
   .....
   chip = kzalloc(sizeof(*chip), GFP_KERNEL);
 ]]>
@@ -1170,8 +1177,6 @@
                   return err;
           }
 
-          snd_card_set_dev(card, &pci->dev);
-
           *rchip = chip;
           return 0;
   }        
@@ -1526,30 +1531,6 @@
 
     </section>
 
-    <section id="pci-resource-device-struct">
-      <title>Registration of Device Struct</title>
-      <para>
-	At some point, typically after calling <function>snd_device_new()</function>,
-	you need to register the struct <structname>device</structname> of the chip
-	you're handling for udev and co.  ALSA provides a macro for compatibility with
-	older kernels.  Simply call like the following:
-        <informalexample>
-          <programlisting>
-<![CDATA[
-  snd_card_set_dev(card, &pci->dev);
-]]>
-          </programlisting>
-        </informalexample>
-	so that it stores the PCI's device pointer to the card.  This will be
-	referred by ALSA core functions later when the devices are registered.
-      </para>
-      <para>
-	In the case of non-PCI, pass the proper device struct pointer of the BUS
-	instead.  (In the case of legacy ISA without PnP, you don't have to do
-	anything.)
-      </para>
-    </section>
-
     <section id="pci-resource-entries">
       <title>PCI Entries</title>
       <para>
@@ -5740,7 +5721,8 @@ struct _snd_pcm_runtime {
           struct mychip *chip;
           int err;
           ....
-          err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+          err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+                             0, &card);
           ....
           chip = kzalloc(sizeof(*chip), GFP_KERNEL);
           ....
@@ -5752,7 +5734,7 @@ struct _snd_pcm_runtime {
       </informalexample>
 
 	When you created the chip data with
-	<function>snd_card_create()</function>, it's anyway accessible
+	<function>snd_card_new()</function>, it's anyway accessible
 	via <structfield>private_data</structfield> field.
 
       <informalexample>
@@ -5766,8 +5748,8 @@ struct _snd_pcm_runtime {
           struct mychip *chip;
           int err;
           ....
-          err = snd_card_create(index[dev], id[dev], THIS_MODULE,
-                                sizeof(struct mychip), &card);
+          err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+                             sizeof(struct mychip), &card);
           ....
           chip = card->private_data;
           ....
diff --git a/include/sound/core.h b/include/sound/core.h
index d0cee2c8c04f..e946b2428ea0 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -283,9 +283,16 @@ int snd_card_locked(int card);
 extern int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int cmd);
 #endif
 
-int snd_card_create(int idx, const char *id,
-		    struct module *module, int extra_size,
-		    struct snd_card **card_ret);
+int snd_card_new(struct device *parent, int idx, const char *xid,
+		 struct module *module, int extra_size,
+		 struct snd_card **card_ret);
+
+static inline int __deprecated
+snd_card_create(int idx, const char *id, struct module *module, int extra_size,
+		struct snd_card **ret)
+{
+	return snd_card_new(NULL, idx, id, module, extra_size, ret);
+}
 
 int snd_card_disconnect(struct snd_card *card);
 int snd_card_free(struct snd_card *card);
diff --git a/sound/core/init.c b/sound/core/init.c
index a16d765cdf47..f4d3ac633ff8 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -157,7 +157,8 @@ static int get_slot_from_bitmask(int mask, int (*check)(struct module *, int),
 }
 
 /**
- *  snd_card_create - create and initialize a soundcard structure
+ *  snd_card_new - create and initialize a soundcard structure
+ *  @parent: the parent device object
  *  @idx: card index (address) [0 ... (SNDRV_CARDS-1)]
  *  @xid: card identification (ASCII string)
  *  @module: top level module for locking
@@ -172,7 +173,7 @@ static int get_slot_from_bitmask(int mask, int (*check)(struct module *, int),
  *
  *  Return: Zero if successful or a negative error code.
  */
-int snd_card_create(int idx, const char *xid,
+int snd_card_new(struct device *parent, int idx, const char *xid,
 		    struct module *module, int extra_size,
 		    struct snd_card **card_ret)
 {
@@ -213,6 +214,7 @@ int snd_card_create(int idx, const char *xid,
 	if (idx >= snd_ecards_limit)
 		snd_ecards_limit = idx + 1; /* increase the limit */
 	mutex_unlock(&snd_card_mutex);
+	card->dev = parent;
 	card->number = idx;
 	card->module = module;
 	INIT_LIST_HEAD(&card->devices);
@@ -251,7 +253,7 @@ int snd_card_create(int idx, const char *xid,
 	kfree(card);
   	return err;
 }
-EXPORT_SYMBOL(snd_card_create);
+EXPORT_SYMBOL(snd_card_new);
 
 /* return non-zero if a card is already locked */
 int snd_card_locked(int card)
-- 
cgit v1.2.3


From 86f28d76435b619bd0bc5f6fde2803a5bc27ca24 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 11 Feb 2014 15:42:48 +0800
Subject: ASoC: fsl-spdif: big-endian support

For most platforms, the CPU and SPDIF device is in the same endianess
mode. While for the LS1 platform, the CPU is in LE mode and the SPDIF
is in BE mode.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/fsl,spdif.txt | 5 +++++
 sound/soc/fsl/fsl_spdif.c                             | 5 ++++-
 2 files changed, 9 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
index f2ae335670f5..3e9e82c8eab3 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
@@ -29,6 +29,10 @@ Required properties:
 			can also be referred to TxClk_Source
 			bit of register SPDIF_STC.
 
+   - big-endian : If this property is absent, the native endian mode will
+   be in use as default, or the big endian mode will be in use for all the
+   device registers.
+
 Example:
 
 spdif: spdif@02004000 {
@@ -50,5 +54,6 @@ spdif: spdif@02004000 {
 		"rxtx5", "rxtx6",
 		"rxtx7";
 
+	big-endian;
 	status = "okay";
 };
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 4d075f1abe78..73ceb2f9b90d 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -985,7 +985,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static const struct regmap_config fsl_spdif_regmap_config = {
+static struct regmap_config fsl_spdif_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1105,6 +1105,9 @@ 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);
-- 
cgit v1.2.3


From eaba603fc7c6281908c316d9e58de688943d58be Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 11 Feb 2014 15:42:49 +0800
Subject: ASoC: fsl-esai: big-endian support

For most platforms, the CPU and ESAI device is in the same endianess
mode. While for the LS1 platform, the CPU is in LE mode and the ESAI
is in BE mode.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/fsl,esai.txt | 5 +++++
 sound/soc/fsl/fsl_esai.c                             | 5 ++++-
 2 files changed, 9 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
index d7b99fa637b5..aeb8c4a0b88d 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -34,6 +34,10 @@ Required properties:
     that ESAI would work in the synchronous mode, which means all the settings
     for Receiving would be duplicated from Transmition related registers.
 
+  - big-endian : If this property is absent, the native endian mode will
+    be in use as default, or the big endian mode will be in use for all the
+    device registers.
+
 Example:
 
 esai: esai@02024000 {
@@ -46,5 +50,6 @@ esai: esai@02024000 {
 	dma-names = "rx", "tx";
 	fsl,fifo-depth = <128>;
 	fsl,esai-synchronous;
+	big-endian;
 	status = "disabled";
 };
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index f55341e52970..d8e13abd1bca 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -678,7 +678,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static const struct regmap_config fsl_esai_regmap_config = {
+static struct regmap_config fsl_esai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -704,6 +704,9 @@ 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);
-- 
cgit v1.2.3


From a69d0009aad51e16bf294d78a3b2e4fe655cf10f Mon Sep 17 00:00:00 2001
From: Bo Shen <voice.shen@atmel.com>
Date: Mon, 10 Feb 2014 14:09:46 +0800
Subject: Binding: atmel-ssc: add option to choose clock

Add the option to choose clock on which pin input to SSC (as slave).
Default is on TK pin to SSC, add "atmel,clk-from-rk-pin" option to
specify the clock is on RK pin to SSC.

Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/misc/atmel-ssc.txt | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/misc/atmel-ssc.txt b/Documentation/devicetree/bindings/misc/atmel-ssc.txt
index 60960b2755f4..efc98ea1f23d 100644
--- a/Documentation/devicetree/bindings/misc/atmel-ssc.txt
+++ b/Documentation/devicetree/bindings/misc/atmel-ssc.txt
@@ -17,6 +17,14 @@ Required properties for devices compatible with "atmel,at91sam9g45-ssc":
   See Documentation/devicetree/bindings/dma/atmel-dma.txt for details.
 - dma-names: Must be "tx", "rx".
 
+Optional properties:
+  - atmel,clk-from-rk-pin: bool property.
+     - When SSC works in slave mode, according to the hardware design, the
+       clock can get from TK pin, and also can get from RK pin. So, add
+       this parameter to choose where the clock from.
+     - By default the clock is from TK pin, if the clock from RK pin, this
+       property is needed.
+
 Examples:
 - PDC transfer:
 ssc0: ssc@fffbc000 {
-- 
cgit v1.2.3


From 74d04c3efbc4f10990e5c4218ad3f65bfdcf3c75 Mon Sep 17 00:00:00 2001
From: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Date: Wed, 12 Feb 2014 18:20:56 +0100
Subject: sound: ASoC: add ASoC board driver for Armada 370 DB

This commit adds a simple ASoC board driver fo the Armada 370
Development Board, which connects the audio unit of the Armada 370 SoC
to the I2C-based CS42L51.

For now, only the analog audio input and output through the CS42L51
are supported, but a followup patch adds S/PDIF support to this
driver.

Signed-off-by: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../bindings/sound/armada-370db-audio.txt          |  24 +++++
 sound/soc/kirkwood/Kconfig                         |   8 ++
 sound/soc/kirkwood/Makefile                        |   2 +
 sound/soc/kirkwood/armada-370-db.c                 | 120 +++++++++++++++++++++
 4 files changed, 154 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt
 create mode 100644 sound/soc/kirkwood/armada-370-db.c

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
new file mode 100644
index 000000000000..3893b4d15a20
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
@@ -0,0 +1,24 @@
+Device Tree bindings for the Armada 370 DB audio
+================================================
+
+These Device Tree bindings are used to describe the audio complex
+found on the Armada 370 DB platform.
+
+Mandatory properties:
+
+ * compatible: must be "marvell,a370db-audio"
+
+ * marvell,audio-controller: a phandle that points to the audio
+   controller of the Armada 370 SoC.
+
+ * marvell,audio-codec: a phandle that points to the analog audio
+   codec connected to the Armada 370 SoC.
+
+Example:
+
+	sound {
+	      compatible = "marvell,a370db-audio";
+	      marvell,audio-controller = <&audio_controller>;
+	      marvell,audio-codec = <&audio_codec>;
+	      status = "okay";
+	};
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 764a0ef6b268..2dc3ecf34801 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -6,6 +6,14 @@ config SND_KIRKWOOD_SOC
 	  the Kirkwood I2S interface. You will also need to select the
 	  audio interfaces to support below.
 
+config SND_KIRKWOOD_SOC_ARMADA370_DB
+	tristate "SoC Audio support for Armada 370 DB"
+	depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C
+	select SND_SOC_CS42L51
+	help
+	  Say Y if you want to add support for SoC audio on
+	  the Armada 370 Development Board.
+
 config SND_KIRKWOOD_SOC_OPENRD
 	tristate "SoC Audio support for Kirkwood Openrd Client"
 	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 9e781385cb88..7c1d8fe09e6b 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -4,6 +4,8 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
 
 snd-soc-openrd-objs := kirkwood-openrd.o
 snd-soc-t5325-objs := kirkwood-t5325.o
+snd-soc-armada-370-db-objs := armada-370-db.o
 
+obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
 obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
 obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
new file mode 100644
index 000000000000..977639b3ffde
--- /dev/null
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -0,0 +1,120 @@
+/*
+ * Copyright (C) 2014 Marvell
+ *
+ * Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
+ *
+ * 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; either version 2 of the
+ * License, or (at your option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <linux/of.h>
+#include <linux/platform_data/asoc-kirkwood.h>
+#include "../codecs/cs42l51.h"
+
+static int a370db_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;
+	unsigned int freq;
+
+	switch (params_rate(params)) {
+	default:
+	case 44100:
+		freq = 11289600;
+		break;
+	case 48000:
+		freq = 12288000;
+		break;
+	case 96000:
+		freq = 24576000;
+		break;
+	}
+
+	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops a370db_ops = {
+	.hw_params = a370db_hw_params,
+};
+
+static const struct snd_soc_dapm_widget a370db_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Out Jack", NULL),
+	SND_SOC_DAPM_LINE("In Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route a370db_route[] = {
+	{ "Out Jack",	NULL,	"HPL" },
+	{ "Out Jack",	NULL,	"HPR" },
+	{ "AIN1L",	NULL,	"In Jack" },
+	{ "AIN1L",	NULL,	"In Jack" },
+};
+
+static struct snd_soc_dai_link a370db_dai[] = {
+{
+	.name = "CS42L51",
+	.stream_name = "analog",
+	.cpu_dai_name = "i2s",
+	.codec_dai_name = "cs42l51-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &a370db_ops,
+},
+};
+
+static struct snd_soc_card a370db = {
+	.name = "a370db",
+	.owner = THIS_MODULE,
+	.dai_link = a370db_dai,
+	.num_links = ARRAY_SIZE(a370db_dai),
+	.dapm_widgets = a370db_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(a370db_dapm_widgets),
+	.dapm_routes = a370db_route,
+	.num_dapm_routes = ARRAY_SIZE(a370db_route),
+};
+
+static int a370db_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &a370db;
+
+	card->dev = &pdev->dev;
+
+	a370db_dai[0].cpu_of_node =
+		of_parse_phandle(pdev->dev.of_node,
+				 "marvell,audio-controller", 0);
+	a370db_dai[0].platform_of_node = a370db_dai[0].cpu_of_node;
+
+	a370db_dai[0].codec_of_node =
+		of_parse_phandle(pdev->dev.of_node,
+				 "marvell,audio-codec", 0);
+
+	return devm_snd_soc_register_card(card->dev, card);
+}
+
+static const struct of_device_id a370db_dt_ids[] = {
+	{ .compatible = "marvell,a370db-audio" },
+	{ },
+};
+
+static struct platform_driver a370db_driver = {
+	.driver		= {
+		.name	= "a370db-audio",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(a370db_dt_ids),
+	},
+	.probe		= a370db_probe,
+};
+
+module_platform_driver(a370db_driver);
+
+MODULE_AUTHOR("Thomas Petazzoni <thomas.petazzoni@free-electrons.com>");
+MODULE_DESCRIPTION("ALSA SoC a370db audio client");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:a370db-audio");
-- 
cgit v1.2.3


From 98b664e2ceddd40120e8cd2aa56f7eb9a51870cf Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Thu, 20 Feb 2014 18:22:58 +0100
Subject: ASoC: tlv320aic32x4: Support for master clock

Add support for a master clock passed through DT. The master clock of
the codec is only active when the codec is in use.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/tlv320aic32x4.txt     |  4 ++++
 sound/soc/codecs/tlv320aic32x4.c                    | 21 +++++++++++++++++++++
 2 files changed, 25 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
index db0551088cc4..352be7b1f7e2 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -8,6 +8,8 @@ Required properties:
 
 Optional properties:
  - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
+ - clocks/clock-names: Clock named 'mclk' for the master clock of the codec.
+   See clock/clock-bindings.txt for information about the detailed format.
 
 
 Example:
@@ -15,4 +17,6 @@ Example:
 codec: tlv320aic32x4@18 {
 	compatible = "ti,tlv320aic32x4";
 	reg = <0x18>;
+	clocks = <&clks 201>;
+	clock-names = "mclk";
 };
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 1dd50e48934c..643fa53beaab 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -33,6 +33,7 @@
 #include <linux/i2c.h>
 #include <linux/cdev.h>
 #include <linux/slab.h>
+#include <linux/clk.h>
 
 #include <sound/tlv320aic32x4.h>
 #include <sound/core.h>
@@ -67,6 +68,7 @@ struct aic32x4_priv {
 	u32 micpga_routing;
 	bool swapdacs;
 	int rstn_gpio;
+	struct clk *mclk;
 };
 
 /* 0dB min, 0.5dB steps */
@@ -487,8 +489,18 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
 static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
+		/* Switch on master clock */
+		ret = clk_prepare_enable(aic32x4->mclk);
+		if (ret) {
+			dev_err(codec->dev, "Failed to enable master clock\n");
+			return ret;
+		}
+
 		/* Switch on PLL */
 		snd_soc_update_bits(codec, AIC32X4_PLLPR,
 				    AIC32X4_PLLEN, AIC32X4_PLLEN);
@@ -539,6 +551,9 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
 		/* Switch off BCLK_N Divider */
 		snd_soc_update_bits(codec, AIC32X4_BCLKN,
 				    AIC32X4_BCLKEN, 0);
+
+		/* Switch off master clock */
+		clk_disable_unprepare(aic32x4->mclk);
 		break;
 	case SND_SOC_BIAS_OFF:
 		break;
@@ -717,6 +732,12 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
 		aic32x4->rstn_gpio = -1;
 	}
 
+	aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk");
+	if (IS_ERR(aic32x4->mclk)) {
+		dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
+		return PTR_ERR(aic32x4->mclk);
+	}
+
 	if (gpio_is_valid(aic32x4->rstn_gpio)) {
 		ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
 				GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
-- 
cgit v1.2.3


From 239b669b2dedc46d5e6b07d87c3d1dedf8d9477c Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Thu, 20 Feb 2014 18:22:59 +0100
Subject: ASoC: tlv320aic32x4: Support for regulators

Support regulators to power up the codec. This patch also enables the
AVDD LDO if no AV regulator was found.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/tlv320aic32x4.txt    |   8 ++
 sound/soc/codecs/tlv320aic32x4.c                   | 126 ++++++++++++++++++++-
 2 files changed, 133 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
index 352be7b1f7e2..5e2741af27be 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -5,6 +5,14 @@ The tlv320aic32x4 serial control bus communicates through I2C protocols
 Required properties:
  - compatible: Should be "ti,tlv320aic32x4"
  - reg: I2C slave address
+ - supply-*: Required supply regulators are:
+    "iov" - digital IO power supply
+    "ldoin" - LDO power supply
+    "dv" - Digital core power supply
+    "av" - Analog core power supply
+    If you supply ldoin, dv and av are optional. Otherwise they are required
+   See regulator/regulator.txt for more information about the detailed binding
+   format.
 
 Optional properties:
  - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 643fa53beaab..d69c61ffcda8 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -34,6 +34,7 @@
 #include <linux/cdev.h>
 #include <linux/slab.h>
 #include <linux/clk.h>
+#include <linux/regulator/consumer.h>
 
 #include <sound/tlv320aic32x4.h>
 #include <sound/core.h>
@@ -69,6 +70,11 @@ struct aic32x4_priv {
 	bool swapdacs;
 	int rstn_gpio;
 	struct clk *mclk;
+
+	struct regulator *supply_ldo;
+	struct regulator *supply_iov;
+	struct regulator *supply_dv;
+	struct regulator *supply_av;
 };
 
 /* 0dB min, 0.5dB steps */
@@ -695,6 +701,106 @@ static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4,
 	return 0;
 }
 
+static void aic32x4_disable_regulators(struct aic32x4_priv *aic32x4)
+{
+	regulator_disable(aic32x4->supply_iov);
+
+	if (!IS_ERR(aic32x4->supply_ldo))
+		regulator_disable(aic32x4->supply_ldo);
+
+	if (!IS_ERR(aic32x4->supply_dv))
+		regulator_disable(aic32x4->supply_dv);
+
+	if (!IS_ERR(aic32x4->supply_av))
+		regulator_disable(aic32x4->supply_av);
+}
+
+static int aic32x4_setup_regulators(struct device *dev,
+		struct aic32x4_priv *aic32x4)
+{
+	int ret = 0;
+
+	aic32x4->supply_ldo = devm_regulator_get_optional(dev, "ldoin");
+	aic32x4->supply_iov = devm_regulator_get(dev, "iov");
+	aic32x4->supply_dv = devm_regulator_get_optional(dev, "dv");
+	aic32x4->supply_av = devm_regulator_get_optional(dev, "av");
+
+	/* Check if the regulator requirements are fulfilled */
+
+	if (IS_ERR(aic32x4->supply_iov)) {
+		dev_err(dev, "Missing supply 'iov'\n");
+		return PTR_ERR(aic32x4->supply_iov);
+	}
+
+	if (IS_ERR(aic32x4->supply_ldo)) {
+		if (PTR_ERR(aic32x4->supply_ldo) == -EPROBE_DEFER)
+			return -EPROBE_DEFER;
+
+		if (IS_ERR(aic32x4->supply_dv)) {
+			dev_err(dev, "Missing supply 'dv' or 'ldoin'\n");
+			return PTR_ERR(aic32x4->supply_dv);
+		}
+		if (IS_ERR(aic32x4->supply_av)) {
+			dev_err(dev, "Missing supply 'av' or 'ldoin'\n");
+			return PTR_ERR(aic32x4->supply_av);
+		}
+	} else {
+		if (IS_ERR(aic32x4->supply_dv) &&
+				PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER)
+			return -EPROBE_DEFER;
+		if (IS_ERR(aic32x4->supply_av) &&
+				PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER)
+			return -EPROBE_DEFER;
+	}
+
+	ret = regulator_enable(aic32x4->supply_iov);
+	if (ret) {
+		dev_err(dev, "Failed to enable regulator iov\n");
+		return ret;
+	}
+
+	if (!IS_ERR(aic32x4->supply_ldo)) {
+		ret = regulator_enable(aic32x4->supply_ldo);
+		if (ret) {
+			dev_err(dev, "Failed to enable regulator ldo\n");
+			goto error_ldo;
+		}
+	}
+
+	if (!IS_ERR(aic32x4->supply_dv)) {
+		ret = regulator_enable(aic32x4->supply_dv);
+		if (ret) {
+			dev_err(dev, "Failed to enable regulator dv\n");
+			goto error_dv;
+		}
+	}
+
+	if (!IS_ERR(aic32x4->supply_av)) {
+		ret = regulator_enable(aic32x4->supply_av);
+		if (ret) {
+			dev_err(dev, "Failed to enable regulator av\n");
+			goto error_av;
+		}
+	}
+
+	if (!IS_ERR(aic32x4->supply_ldo) && IS_ERR(aic32x4->supply_av))
+		aic32x4->power_cfg |= AIC32X4_PWR_AIC32X4_LDO_ENABLE;
+
+	return 0;
+
+error_av:
+	if (!IS_ERR(aic32x4->supply_dv))
+		regulator_disable(aic32x4->supply_dv);
+
+error_dv:
+	if (!IS_ERR(aic32x4->supply_ldo))
+		regulator_disable(aic32x4->supply_ldo);
+
+error_ldo:
+	regulator_disable(aic32x4->supply_iov);
+	return ret;
+}
+
 static int aic32x4_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
@@ -745,13 +851,31 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
 			return ret;
 	}
 
+	ret = aic32x4_setup_regulators(&i2c->dev, aic32x4);
+	if (ret) {
+		dev_err(&i2c->dev, "Failed to setup regulators\n");
+		return ret;
+	}
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
-	return ret;
+	if (ret) {
+		dev_err(&i2c->dev, "Failed to register codec\n");
+		aic32x4_disable_regulators(aic32x4);
+		return ret;
+	}
+
+	i2c_set_clientdata(i2c, aic32x4);
+
+	return 0;
 }
 
 static int aic32x4_i2c_remove(struct i2c_client *client)
 {
+	struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client);
+
+	aic32x4_disable_regulators(aic32x4);
+
 	snd_soc_unregister_codec(&client->dev);
 	return 0;
 }
-- 
cgit v1.2.3


From 72899ad8a46a80ad5fc725afea443f94945ae370 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 14 Feb 2014 09:34:34 +0800
Subject: ASoC: binding: add tdm-slot.txt

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/tdm-slot.txt | 20 ++++++++++++++++++++
 1 file changed, 20 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt
new file mode 100644
index 000000000000..6a2c84247f91
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt
@@ -0,0 +1,20 @@
+TDM slot:
+
+This specifies audio DAI's TDM slot.
+
+TDM slot properties:
+dai-tdm-slot-num : Number of slots in use.
+dai-tdm-slot-width :  Width in bits for each slot.
+
+For instance:
+	dai-tdm-slot-num = <2>;
+	dai-tdm-slot-width = <8>;
+
+And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
+to specify a explicit mapping of the channels and the slots. If it's absent
+the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
+tx and rx masks.
+
+For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
+for an active slot as default, and the default active bits are at the LSB of
+the masks.
-- 
cgit v1.2.3


From 6ff62eedce4f7756b092d276165d8e11edab9f28 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 14 Feb 2014 09:34:36 +0800
Subject: ASoC: simple-card: add slot information parsing supports

For some CPU/CODEC DAI devices the slot information maybe needed. This
patch adds the slot information parsing for simple-card driver.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/simple-card.txt          |  5 +++++
 include/sound/simple_card.h                            |  2 ++
 sound/soc/generic/simple-card.c                        | 18 ++++++++++++++++++
 3 files changed, 25 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index 05273583490c..b30c222f9cd3 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -18,6 +18,8 @@ Optional properties:
 					  Each entry is a pair of strings, the first being the
 					  connection's sink, the second being the connection's
 					  source.
+- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
+- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
 
 Required subnodes:
 
@@ -56,6 +58,9 @@ sound {
 		"Headphone Jack", "HP_OUT",
 		"External Speaker", "LINE_OUT";
 
+	dai-tdm-slot-num = <2>;
+	dai-tdm-slot-width = <8>;
+
 	simple-audio-card,cpu {
 		sound-dai = <&sh_fsi2 0>;
 	};
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
index e1ac996c8feb..9b0ac77177b6 100644
--- a/include/sound/simple_card.h
+++ b/include/sound/simple_card.h
@@ -18,6 +18,8 @@ struct asoc_simple_dai {
 	const char *name;
 	unsigned int fmt;
 	unsigned int sysclk;
+	int slots;
+	int slot_width;
 };
 
 struct asoc_simple_card_info {
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 4fe8abc6e216..bdd176ddff07 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -9,11 +9,14 @@
  * published by the Free Software Foundation.
  */
 #include <linux/clk.h>
+#include <linux/device.h>
 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/platform_device.h>
 #include <linux/string.h>
 #include <sound/simple_card.h>
+#include <sound/soc-dai.h>
+#include <sound/soc.h>
 
 struct simple_card_data {
 	struct snd_soc_card snd_card;
@@ -44,6 +47,16 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
 		}
 	}
 
+	if (set->slots) {
+		ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+						set->slots,
+						set->slot_width);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(dai->dev, "simple-card: set_tdm_slot error\n");
+			goto err;
+		}
+	}
+
 	ret = 0;
 
 err:
@@ -94,6 +107,11 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 	if (ret < 0)
 		goto parse_error;
 
+	/* parse TDM slot */
+	ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+	if (ret)
+		goto parse_error;
+
 	/*
 	 * bitclock-inversion, frame-inversion
 	 * bitclock-master,    frame-master
-- 
cgit v1.2.3


From ac5630b504be8918f42f4bd62f75063e469adf8b Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Wed, 26 Feb 2014 17:14:33 +0100
Subject: ASoC: tlv320aic3x: Remove tlv320aic32x4 from compatibles of
 tlv320aic3x

This reverts tlv320aic32x4 as compatible for tlv320aic3x as it has its
own bindings now.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 -
 1 file changed, 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index 9d8ea14db490..5e6040c2c2e9 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -6,7 +6,6 @@ Required properties:
 
 - compatible - "string" - One of:
     "ti,tlv320aic3x" - Generic TLV320AIC3x device
-    "ti,tlv320aic32x4" - TLV320AIC32x4
     "ti,tlv320aic33" - TLV320AIC33
     "ti,tlv320aic3007" - TLV320AIC3007
     "ti,tlv320aic3106" - TLV320AIC3106
-- 
cgit v1.2.3


From 66f232908de2f7393a1f7c4c300f73534af57f07 Mon Sep 17 00:00:00 2001
From: Denis Carikli <denis@eukrea.com>
Date: Wed, 15 Jan 2014 18:23:25 +0100
Subject: ASoC: eukrea-tlv320: Add DT support.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Cc: Eric Bénard <eric@eukrea.com>
Cc: Ian Campbell <ijc+devicetree@hellion.org.uk>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Mark Rutland <mark.rutland@arm.com>
Cc: Pawel Moll <pawel.moll@arm.com>
Cc: Rob Herring <robh+dt@kernel.org>
Cc: alsa-devel@alsa-project.org
Cc: devicetree@vger.kernel.org
Signed-off-by: Denis Carikli <denis@eukrea.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/eukrea-tlv320.txt    |  21 ++++
 sound/soc/fsl/Kconfig                              |   5 +-
 sound/soc/fsl/eukrea-tlv320.c                      | 108 ++++++++++++++++++---
 3 files changed, 121 insertions(+), 13 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
new file mode 100644
index 000000000000..0d7985c864af
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
@@ -0,0 +1,21 @@
+Audio complex for Eukrea boards with tlv320aic23 codec.
+
+Required properties:
+- compatible : "eukrea,asoc-tlv320"
+- eukrea,model : The user-visible name of this sound complex.
+- ssi-controller : The phandle of the SSI controller.
+- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX).
+- fsl,mux-ext-port : The external port of the i.MX audio muxer.
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+	sound {
+		compatible = "eukrea,asoc-tlv320";
+		eukrea,model = "imx51-eukrea-tlv320aic23";
+		ssi-controller = <&ssi2>;
+		fsl,mux-int-port = <2>;
+		fsl,mux-ext-port = <3>;
+	};
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 07f8f141727d..b8f8703e2b46 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -168,12 +168,15 @@ config SND_SOC_EUKREA_TLV320
 	depends on MACH_EUKREA_MBIMX27_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
-		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
+		|| MACH_EUKREA_MBIMXSD51_BASEBOARD \
+		|| (OF && ARM)
 	depends on I2C
 	select SND_SOC_TLV320AIC23
 	select SND_SOC_IMX_PCM_FIQ
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_SSI
+	select SND_SOC_FSL_SSI
+	select SND_SOC_IMX_PCM_DMA
 	help
 	  Enable I2S based access to the TLV320AIC23B codec attached
 	  to the SSI interface
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 5983740be123..eb093d5b85c4 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -15,8 +15,11 @@
  *
  */
 
+#include <linux/errno.h>
 #include <linux/module.h>
 #include <linux/moduleparam.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
 #include <linux/device.h>
 #include <linux/i2c.h>
 #include <sound/core.h>
@@ -26,6 +29,7 @@
 
 #include "../codecs/tlv320aic23.h"
 #include "imx-ssi.h"
+#include "fsl_ssi.h"
 #include "imx-audmux.h"
 
 #define CODEC_CLOCK 12000000
@@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
 	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
 				  SND_SOC_DAIFMT_NB_NF |
 				  SND_SOC_DAIFMT_CBM_CFM);
-	if (ret) {
+	/* fsl_ssi lacks the set_fmt ops. */
+	if (ret && ret != -ENOTSUPP) {
 		dev_err(cpu_dai->dev,
 			"Failed to set the cpu dai format.\n");
 		return ret;
@@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
 			"Failed to set the codec sysclk.\n");
 		return ret;
 	}
+
 	snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
 
 	ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
 				SND_SOC_CLOCK_IN);
-	if (ret) {
+	/* fsl_ssi lacks the set_sysclk ops */
+	if (ret && ret != -EINVAL) {
 		dev_err(cpu_dai->dev,
 			"Can't set the IMX_SSP_SYS_CLK CPU system clock.\n");
 		return ret;
@@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
 	.codec_dai_name	= "tlv320aic23-hifi",
-	.platform_name	= "imx-ssi.0",
-	.codec_name	= "tlv320aic23-codec.0-001a",
-	.cpu_dai_name	= "imx-ssi.0",
 	.ops		= &eukrea_tlv320_snd_ops,
 };
 
 static struct snd_soc_card eukrea_tlv320 = {
-	.name		= "cpuimx-audio",
 	.owner		= THIS_MODULE,
 	.dai_link	= &eukrea_tlv320_dai,
 	.num_links	= 1,
@@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
 {
 	int ret;
 	int int_port = 0, ext_port;
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *ssi_np, *codec_np;
 
-	if (machine_is_eukrea_cpuimx27()) {
+	eukrea_tlv320.dev = &pdev->dev;
+	if (np) {
+		ret = snd_soc_of_parse_card_name(&eukrea_tlv320,
+						 "eukrea,model");
+		if (ret) {
+			dev_err(&pdev->dev,
+				"eukrea,model node missing or invalid.\n");
+			goto err;
+		}
+
+		ssi_np = of_parse_phandle(pdev->dev.of_node,
+					  "ssi-controller", 0);
+		if (!ssi_np) {
+			dev_err(&pdev->dev,
+				"ssi-controller missing or invalid.\n");
+			ret = -ENODEV;
+			goto err;
+		}
+
+		codec_np = of_parse_phandle(ssi_np, "codec-handle", 0);
+		if (codec_np)
+			eukrea_tlv320_dai.codec_of_node = codec_np;
+		else
+			dev_err(&pdev->dev, "codec-handle node missing or invalid.\n");
+
+		ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port);
+		if (ret) {
+			dev_err(&pdev->dev,
+				"fsl,mux-int-port node missing or invalid.\n");
+			return ret;
+		}
+		ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
+		if (ret) {
+			dev_err(&pdev->dev,
+				"fsl,mux-ext-port node 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--;
+
+		eukrea_tlv320_dai.cpu_of_node = ssi_np;
+		eukrea_tlv320_dai.platform_of_node = ssi_np;
+	} else {
+		eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0";
+		eukrea_tlv320_dai.platform_name = "imx-ssi.0";
+		eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a";
+		eukrea_tlv320.name = "cpuimx-audio";
+	}
+
+	if (machine_is_eukrea_cpuimx27() ||
+	    of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) {
 		imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
 			IMX_AUDMUX_V1_PCR_SYN |
 			IMX_AUDMUX_V1_PCR_TFSDIR |
@@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
 		);
 	} else if (machine_is_eukrea_cpuimx25sd() ||
 		   machine_is_eukrea_cpuimx35sd() ||
-		   machine_is_eukrea_cpuimx51sd()) {
-		ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3;
+		   machine_is_eukrea_cpuimx51sd() ||
+		   of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) {
+		if (!np)
+			ext_port = machine_is_eukrea_cpuimx25sd() ?
+				4 : 3;
+
 		imx_audmux_v2_configure_port(int_port,
 			IMX_AUDMUX_V2_PTCR_SYN |
 			IMX_AUDMUX_V2_PTCR_TFSDIR |
@@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
 			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)
 		);
 	} else {
-		/* return happy. We might run on a totally different machine */
-		return 0;
+		if (np) {
+			/* The eukrea,asoc-tlv320 driver was explicitely
+			 * requested (through the device tree).
+			 */
+			dev_err(&pdev->dev,
+				"Missing or invalid audmux DT node.\n");
+			return -ENODEV;
+		} else {
+			/* Return happy.
+			 * We might run on a totally different machine.
+			 */
+			return 0;
+		}
 	}
 
-	eukrea_tlv320.dev = &pdev->dev;
 	ret = snd_soc_register_card(&eukrea_tlv320);
+err:
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+	if (np)
+		of_node_put(ssi_np);
 
 	return ret;
 }
@@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct of_device_id imx_tlv320_dt_ids[] = {
+	{ .compatible = "eukrea,asoc-tlv320"},
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids);
+
 static struct platform_driver eukrea_tlv320_driver = {
 	.driver = {
 		.name = "eukrea_tlv320",
 		.owner = THIS_MODULE,
+		.of_match_table = imx_tlv320_dt_ids,
 	},
 	.probe = eukrea_tlv320_probe,
 	.remove = eukrea_tlv320_remove,
-- 
cgit v1.2.3


From f516e368dcb5eb5fbe23246c09bf69573d67cd18 Mon Sep 17 00:00:00 2001
From: Rongjun Ying <rongjun.ying@csr.com>
Date: Wed, 5 Mar 2014 16:34:34 +0800
Subject: ASoC: sirf: Add SiRF internal audio codec driver

SiRF internal audio codec is integrated in SiRF atlas6 and prima2 SoC.
Features include:
1. Stereo DAC and ADC with 16-bit resolution amd 48KHz sample rate
2. Support headphone and/or speaker output
3. Integrate headphone and speaker output amp
4. Support LINE and MIC input
5. Support single ended and differential input mode

Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
--v5:
1. Drop all inlines.
2. Reordering the Kconfig and Makefile
3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it.
4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver
5. Add audio codec clock supply instead of adc event callback
6. Fixed playback and capture can't concurrent work bug.

--
 .../devicetree/bindings/sound/sirf-audio-codec.txt |   17 +
 sound/soc/codecs/Kconfig                           |    5 +
 sound/soc/codecs/Makefile                          |    1 +
 sound/soc/codecs/sirf-audio-codec.c                |  533 ++++++++++++++++++++
 sound/soc/codecs/sirf-audio-codec.h                |   75 +++
 5 files changed, 631 insertions(+), 0 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
 create mode 100644 sound/soc/codecs/sirf-audio-codec.c
 create mode 100644 sound/soc/codecs/sirf-audio-codec.h
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/sirf-audio-codec.txt |  17 +
 sound/soc/codecs/Kconfig                           |   5 +
 sound/soc/codecs/Makefile                          |   1 +
 sound/soc/codecs/sirf-audio-codec.c                | 533 +++++++++++++++++++++
 sound/soc/codecs/sirf-audio-codec.h                |  75 +++
 5 files changed, 631 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
 create mode 100644 sound/soc/codecs/sirf-audio-codec.c
 create mode 100644 sound/soc/codecs/sirf-audio-codec.h

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
new file mode 100644
index 000000000000..062f5ec36f9b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
@@ -0,0 +1,17 @@
+SiRF internal audio CODEC
+
+Required properties:
+
+  - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec"
+
+  - reg : the register address of the device.
+
+  - clocks: the clock of SiRF internal audio codec
+
+Example:
+
+audiocodec: audiocodec@b0040000 {
+	compatible = "sirf,atlas6-audio-codec";
+	reg = <0xb0040000 0x10000>;
+	clocks = <&clks 27>;
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 983d087aa92a..bf9b12c1a9c0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_RT5640 if I2C
 	select SND_SOC_SGTL5000 if I2C
 	select SND_SOC_SI476X if MFD_SI476X_CORE
+	select SND_SOC_SIRF_AUDIO_CODEC
 	select SND_SOC_SN95031 if INTEL_SCU_IPC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2518 if I2C
@@ -330,6 +331,10 @@ config SND_SOC_SIGMADSP
 	tristate
 	select CRC32
 
+config SND_SOC_SIRF_AUDIO_CODEC
+	tristate "SiRF SoC internal audio codec"
+	select REGMAP_MMIO
+
 config SND_SOC_SN95031
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bc126764a44d..de6d7f81b5f6 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -53,6 +53,7 @@ snd-soc-alc5623-objs := alc5623.o
 snd-soc-alc5632-objs := alc5632.o
 snd-soc-sigmadsp-objs := sigmadsp.o
 snd-soc-si476x-objs := si476x.o
+snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
 snd-soc-sn95031-objs := sn95031.o
 snd-soc-spdif-tx-objs := spdif_transmitter.o
 snd-soc-spdif-rx-objs := spdif_receiver.o
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
new file mode 100644
index 000000000000..90e3a228bae4
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -0,0 +1,533 @@
+/*
+ * SiRF audio codec driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-codec.h"
+
+struct sirf_audio_codec {
+	struct clk *clk;
+	struct regmap *regmap;
+	u32 reg_ctrl0, reg_ctrl1;
+};
+
+static const char * const input_mode_mux[] = {"Single-ended",
+	"Differential"};
+
+static const struct soc_enum input_mode_mux_enum =
+	SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux);
+
+static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control =
+	SOC_DAPM_ENUM("Route", input_mode_mux_enum);
+
+static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0);
+static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0);
+static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6,
+	0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0),
+	0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0),
+);
+
+static struct snd_kcontrol_new volume_controls_atlas6[] = {
+	SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+			0x7F, 0, playback_vol_tlv),
+	SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10,
+			0x3F, 0, capture_vol_tlv_atlas6),
+};
+
+static struct snd_kcontrol_new volume_controls_prima2[] = {
+	SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+			0x7F, 0, playback_vol_tlv),
+	SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10,
+			0x1F, 0, capture_vol_tlv_prima2),
+};
+
+static struct snd_kcontrol_new left_input_path_controls[] = {
+	SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0),
+	SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0),
+};
+
+static struct snd_kcontrol_new right_input_path_controls[] = {
+	SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0),
+	SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0),
+};
+
+static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0);
+
+/* After enable adc, Delay 200ms to avoid pop noise */
+static int adc_enable_delay_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		msleep(200);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static void enable_and_reset_codec(struct regmap *regmap,
+		u32 codec_enable_bits, u32 codec_reset_bits)
+{
+	regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+			codec_enable_bits | codec_reset_bits,
+			codec_enable_bits | ~codec_reset_bits);
+	msleep(20);
+	regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+			codec_reset_bits, codec_reset_bits);
+}
+
+static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+#define ATLAS6_CODEC_ENABLE_BITS (1 << 29)
+#define ATLAS6_CODEC_RESET_BITS (1 << 28)
+	struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		enable_and_reset_codec(sirf_audio_codec->regmap,
+			ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS,
+			~ATLAS6_CODEC_ENABLE_BITS);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+#define PRIMA2_CODEC_ENABLE_BITS (1 << 27)
+#define PRIMA2_CODEC_RESET_BITS (1 << 26)
+	struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		enable_and_reset_codec(sirf_audio_codec->regmap,
+			PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS,
+			~PRIMA2_CODEC_ENABLE_BITS);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+			25, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+			26, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+			27, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+			23, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+			24, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+			25, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget =
+	SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+			atlas6_codec_enable_and_reset_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget =
+	SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+			prima2_codec_enable_and_reset_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0),
+	SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0),
+	SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0,
+			&left_dac_to_hp_left_amp_switch_control),
+	SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0,
+			&left_dac_to_hp_right_amp_switch_control),
+	SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0,
+			&right_dac_to_hp_left_amp_switch_control),
+	SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0,
+			&right_dac_to_hp_right_amp_switch_control),
+	SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+			NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+			&left_dac_to_speaker_lineout_switch_control),
+	SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+			&right_dac_to_speaker_lineout_switch_control),
+	SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("SPKOUT"),
+
+	SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0,
+			adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0,
+			adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0,
+		&left_input_path_controls[0],
+		ARRAY_SIZE(left_input_path_controls)),
+	SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0,
+		&right_input_path_controls[0],
+		ARRAY_SIZE(right_input_path_controls)),
+
+	SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0,
+			&sirf_audio_codec_input_mode_control),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0),
+	SND_SOC_DAPM_INPUT("MICIN1"),
+	SND_SOC_DAPM_INPUT("MICIN2"),
+	SND_SOC_DAPM_INPUT("LINEIN1"),
+	SND_SOC_DAPM_INPUT("LINEIN2"),
+
+	SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0,
+			30, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route sirf_audio_codec_map[] = {
+	{"SPKOUT", NULL, "Speaker Driver"},
+	{"Speaker Driver", NULL, "Speaker amp driver"},
+	{"Speaker amp driver", NULL, "Left dac to speaker lineout"},
+	{"Speaker amp driver", NULL, "Right dac to speaker lineout"},
+	{"Left dac to speaker lineout", "Switch", "DAC left"},
+	{"Right dac to speaker lineout", "Switch", "DAC right"},
+	{"HPOUTL", NULL, "HP Left Driver"},
+	{"HPOUTR", NULL, "HP Right Driver"},
+	{"HP Left Driver", NULL, "HP amp left driver"},
+	{"HP Right Driver", NULL, "HP amp right driver"},
+	{"HP amp left driver", NULL, "Right dac to hp left amp"},
+	{"HP amp right driver", NULL , "Right dac to hp right amp"},
+	{"HP amp left driver", NULL, "Left dac to hp left amp"},
+	{"HP amp right driver", NULL , "Right dac to hp right amp"},
+	{"Right dac to hp left amp", "Switch", "DAC left"},
+	{"Right dac to hp right amp", "Switch", "DAC right"},
+	{"Left dac to hp left amp", "Switch", "DAC left"},
+	{"Left dac to hp right amp", "Switch", "DAC right"},
+	{"DAC left", NULL, "codecclk"},
+	{"DAC right", NULL, "codecclk"},
+	{"DAC left", NULL, "Playback"},
+	{"DAC right", NULL, "Playback"},
+	{"DAC left", NULL, "HSL Phase Opposite"},
+	{"DAC right", NULL, "HSL Phase Opposite"},
+
+	{"Capture", NULL, "ADC left"},
+	{"Capture", NULL, "ADC right"},
+	{"ADC left", NULL, "codecclk"},
+	{"ADC right", NULL, "codecclk"},
+	{"ADC left", NULL, "Left PGA mixer"},
+	{"ADC right", NULL, "Right PGA mixer"},
+	{"Left PGA mixer", "Line Left Switch", "LINEIN2"},
+	{"Right PGA mixer", "Line Right Switch", "LINEIN1"},
+	{"Left PGA mixer", "Mic Left Switch", "MICIN2"},
+	{"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"},
+	{"Mic input mode mux", "Single-ended", "MICIN1"},
+	{"Mic input mode mux", "Differential", "MICIN1"},
+};
+
+static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream,
+		int cmd,
+		struct snd_soc_dai *dai)
+{
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	struct snd_soc_codec *codec = dai->codec;
+	u32 val = 0;
+
+	/*
+	 * This is a workaround, When stop playback,
+	 * need disable HP amp, avoid the current noise.
+	 */
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (playback)
+			val = IC_HSLEN | IC_HSREN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (playback)
+		snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0,
+			IC_HSLEN | IC_HSREN, val);
+	return 0;
+}
+
+struct snd_soc_dai_ops sirf_audio_codec_dai_ops = {
+	.trigger = sirf_audio_codec_trigger,
+};
+
+struct snd_soc_dai_driver sirf_audio_codec_dai = {
+	.name = "sirf-audio-codec",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &sirf_audio_codec_dai_ops,
+};
+
+static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
+
+	pm_runtime_enable(codec->dev);
+	codec->control_data = sirf_audio_codec->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) {
+		snd_soc_dapm_new_controls(dapm,
+			prima2_output_driver_dapm_widgets,
+			ARRAY_SIZE(prima2_output_driver_dapm_widgets));
+		snd_soc_dapm_new_controls(dapm,
+			&prima2_codec_clock_dapm_widget, 1);
+		return snd_soc_add_codec_controls(codec,
+			volume_controls_prima2,
+			ARRAY_SIZE(volume_controls_prima2));
+	}
+	if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) {
+		snd_soc_dapm_new_controls(dapm,
+			atlas6_output_driver_dapm_widgets,
+			ARRAY_SIZE(atlas6_output_driver_dapm_widgets));
+		snd_soc_dapm_new_controls(dapm,
+			&atlas6_codec_clock_dapm_widget, 1);
+		return snd_soc_add_codec_controls(codec,
+			volume_controls_atlas6,
+			ARRAY_SIZE(volume_controls_atlas6));
+	}
+
+	return -EINVAL;
+}
+
+static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
+{
+	pm_runtime_disable(codec->dev);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
+	.probe = sirf_audio_codec_probe,
+	.remove = sirf_audio_codec_remove,
+	.dapm_widgets = sirf_audio_codec_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
+	.dapm_routes = sirf_audio_codec_map,
+	.num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+	.idle_bias_off = true,
+};
+
+static const struct of_device_id sirf_audio_codec_of_match[] = {
+	{ .compatible = "sirf,prima2-audio-codec" },
+	{ .compatible = "sirf,atlas6-audio-codec" },
+	{}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match);
+
+static const struct regmap_config sirf_audio_codec_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = AUDIO_IC_CODEC_CTRL3,
+	.cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct sirf_audio_codec *sirf_audio_codec;
+	void __iomem *base;
+	struct resource *mem_res;
+	const struct of_device_id *match;
+
+	match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node);
+
+	sirf_audio_codec = devm_kzalloc(&pdev->dev,
+		sizeof(struct sirf_audio_codec), GFP_KERNEL);
+	if (!sirf_audio_codec)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, sirf_audio_codec);
+
+	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	if (base == NULL)
+		return -ENOMEM;
+
+	sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+					    &sirf_audio_codec_regmap_config);
+	if (IS_ERR(sirf_audio_codec->regmap))
+		return PTR_ERR(sirf_audio_codec->regmap);
+
+	sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL);
+	if (IS_ERR(sirf_audio_codec->clk)) {
+		dev_err(&pdev->dev, "Get clock failed.\n");
+		return PTR_ERR(sirf_audio_codec->clk);
+	}
+
+	ret = clk_prepare_enable(sirf_audio_codec->clk);
+	if (ret) {
+		dev_err(&pdev->dev, "Enable clock failed.\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_codec(&(pdev->dev),
+			&soc_codec_device_sirf_audio_codec,
+			&sirf_audio_codec_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "Register Audio Codec dai failed.\n");
+		goto err_clk_put;
+	}
+
+	/*
+	 * Always open charge pump, if not, when the charge pump closed the
+	 * adc will not stable
+	 */
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+		IC_CPFREQ, IC_CPFREQ);
+
+	if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec"))
+		regmap_update_bits(sirf_audio_codec->regmap,
+				AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN);
+	return 0;
+
+err_clk_put:
+	clk_disable_unprepare(sirf_audio_codec->clk);
+	return ret;
+}
+
+static int sirf_audio_codec_driver_remove(struct platform_device *pdev)
+{
+	struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev);
+
+	clk_disable_unprepare(sirf_audio_codec->clk);
+	snd_soc_unregister_codec(&(pdev->dev));
+
+	return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int sirf_audio_codec_suspend(struct device *dev)
+{
+	struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+
+	regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+		&sirf_audio_codec->reg_ctrl0);
+	regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+		&sirf_audio_codec->reg_ctrl1);
+	clk_disable_unprepare(sirf_audio_codec->clk);
+
+	return 0;
+}
+
+static int sirf_audio_codec_resume(struct device *dev)
+{
+	struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_prepare_enable(sirf_audio_codec->clk);
+	if (ret)
+		return ret;
+
+	regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+		sirf_audio_codec->reg_ctrl0);
+	regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+		sirf_audio_codec->reg_ctrl1);
+
+	return 0;
+}
+#endif
+
+static const struct dev_pm_ops sirf_audio_codec_pm_ops = {
+	SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume)
+};
+
+static struct platform_driver sirf_audio_codec_driver = {
+	.driver = {
+		.name = "sirf-audio-codec",
+		.owner = THIS_MODULE,
+		.of_match_table = sirf_audio_codec_of_match,
+		.pm = &sirf_audio_codec_pm_ops,
+	},
+	.probe = sirf_audio_codec_driver_probe,
+	.remove = sirf_audio_codec_driver_remove,
+};
+
+module_platform_driver(sirf_audio_codec_driver);
+
+MODULE_DESCRIPTION("SiRF audio codec driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
new file mode 100644
index 000000000000..d4c187b8e54a
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.h
@@ -0,0 +1,75 @@
+/*
+ * SiRF inner codec controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_CODEC_H
+#define _SIRF_AUDIO_CODEC_H
+
+
+#define AUDIO_IC_CODEC_PWR			(0x00E0)
+#define AUDIO_IC_CODEC_CTRL0			(0x00E4)
+#define AUDIO_IC_CODEC_CTRL1			(0x00E8)
+#define AUDIO_IC_CODEC_CTRL2			(0x00EC)
+#define AUDIO_IC_CODEC_CTRL3			(0x00F0)
+
+#define MICBIASEN		(1 << 3)
+
+#define IC_RDACEN		(1 << 0)
+#define IC_LDACEN		(1 << 1)
+#define IC_HSREN		(1 << 2)
+#define IC_HSLEN		(1 << 3)
+#define IC_SPEN			(1 << 4)
+#define IC_CPEN			(1 << 5)
+
+#define IC_HPRSELR		(1 << 6)
+#define IC_HPLSELR		(1 << 7)
+#define IC_HPRSELL		(1 << 8)
+#define IC_HPLSELL		(1 << 9)
+#define IC_SPSELR		(1 << 10)
+#define IC_SPSELL		(1 << 11)
+
+#define IC_MONOR		(1 << 12)
+#define IC_MONOL		(1 << 13)
+
+#define IC_RXOSRSEL		(1 << 28)
+#define IC_CPFREQ		(1 << 29)
+#define IC_HSINVEN		(1 << 30)
+
+#define IC_MICINREN		(1 << 0)
+#define IC_MICINLEN		(1 << 1)
+#define IC_MICIN1SEL		(1 << 2)
+#define IC_MICIN2SEL		(1 << 3)
+#define IC_MICDIFSEL		(1 << 4)
+#define	IC_LINEIN1SEL		(1 << 5)
+#define	IC_LINEIN2SEL		(1 << 6)
+#define	IC_RADCEN		(1 << 7)
+#define	IC_LADCEN		(1 << 8)
+#define	IC_ALM			(1 << 9)
+
+#define IC_DIGMICEN             (1 << 22)
+#define IC_DIGMICFREQ           (1 << 23)
+#define IC_ADC14B_12            (1 << 24)
+#define IC_FIRDAC_HSL_EN        (1 << 25)
+#define IC_FIRDAC_HSR_EN        (1 << 26)
+#define IC_FIRDAC_LOUT_EN       (1 << 27)
+#define IC_POR                  (1 << 28)
+#define IC_CODEC_CLK_EN         (1 << 29)
+#define IC_HP_3DB_BOOST         (1 << 30)
+
+#define IC_ADC_LEFT_GAIN_SHIFT	16
+#define IC_ADC_RIGHT_GAIN_SHIFT 10
+#define IC_ADC_GAIN_MASK	0x3F
+#define IC_MIC_MAX_GAIN		0x39
+
+#define IC_RXPGAR_MASK		0x3F
+#define IC_RXPGAR_SHIFT		14
+#define IC_RXPGAL_MASK		0x3F
+#define IC_RXPGAL_SHIFT		21
+#define IC_RXPGAR		0x7B
+#define IC_RXPGAL		0x7B
+
+#endif /*__SIRF_AUDIO_CODEC_H*/
-- 
cgit v1.2.3


From a731e217df3a2ee3ef9413153ed7b45e578d8687 Mon Sep 17 00:00:00 2001
From: Rongjun Ying <rongjun.ying@csr.com>
Date: Wed, 5 Mar 2014 16:34:35 +0800
Subject: ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio
 codec

This driver is used by SIRF internal audio codec.
Use dedicated SiRF audio port TXFIFO and RXFIFO
Supports two DMA channels for SiRF audio port TXFIFO and RXFIFO
The audio port like as audio bus such as i2s.

Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/sirf-audio-port.txt  |  20 +++
 sound/soc/Kconfig                                  |   1 +
 sound/soc/Makefile                                 |   1 +
 sound/soc/sirf/Kconfig                             |   8 +
 sound/soc/sirf/Makefile                            |   3 +
 sound/soc/sirf/sirf-audio-port.c                   | 194 +++++++++++++++++++++
 sound/soc/sirf/sirf-audio-port.h                   |  62 +++++++
 7 files changed, 289 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt
 create mode 100644 sound/soc/sirf/Kconfig
 create mode 100644 sound/soc/sirf/Makefile
 create mode 100644 sound/soc/sirf/sirf-audio-port.c
 create mode 100644 sound/soc/sirf/sirf-audio-port.h

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-port.txt b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt
new file mode 100644
index 000000000000..1f66de3c8f00
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt
@@ -0,0 +1,20 @@
+* SiRF SoC audio port
+
+Required properties:
+- compatible: "sirf,audio-port"
+- reg: Base address and size entries:
+- dmas: List of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: Identifier string for each DMA request line in the dmas property.
+  These strings correspond 1:1 with the ordered pairs in dmas.
+
+  One of the DMA channels will be responsible for transmission (should be
+  named "tx") and one for reception (should be named "rx").
+
+Example:
+
+audioport: audioport@b0040000 {
+	compatible = "sirf,audio-port";
+	reg = <0xb0040000 0x10000>;
+	dmas = <&dmac1 3>, <&dmac1 8>;
+	dma-names = "rx", "tx";
+};
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index d62ce483a443..0060b31cc3f3 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -50,6 +50,7 @@ source "sound/soc/pxa/Kconfig"
 source "sound/soc/samsung/Kconfig"
 source "sound/soc/s6000/Kconfig"
 source "sound/soc/sh/Kconfig"
+source "sound/soc/sirf/Kconfig"
 source "sound/soc/spear/Kconfig"
 source "sound/soc/tegra/Kconfig"
 source "sound/soc/txx9/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 62a1822e77bf..5f1df02984f8 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -27,6 +27,7 @@ obj-$(CONFIG_SND_SOC)	+= pxa/
 obj-$(CONFIG_SND_SOC)	+= samsung/
 obj-$(CONFIG_SND_SOC)	+= s6000/
 obj-$(CONFIG_SND_SOC)	+= sh/
+obj-$(CONFIG_SND_SOC)	+= sirf/
 obj-$(CONFIG_SND_SOC)	+= spear/
 obj-$(CONFIG_SND_SOC)	+= tegra/
 obj-$(CONFIG_SND_SOC)	+= txx9/
diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig
new file mode 100644
index 000000000000..75b0344d2151
--- /dev/null
+++ b/sound/soc/sirf/Kconfig
@@ -0,0 +1,8 @@
+config SND_SOC_SIRF
+	tristate "SoC Audio for the SiRF SoC chips"
+	depends on ARCH_SIRF || COMPILE_TEST
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_SIRF_AUDIO_PORT
+	select REGMAP_MMIO
+	tristate
diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile
new file mode 100644
index 000000000000..fb012c852b28
--- /dev/null
+++ b/sound/soc/sirf/Makefile
@@ -0,0 +1,3 @@
+snd-soc-sirf-audio-port-objs := sirf-audio-port.o
+
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o
diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c
new file mode 100644
index 000000000000..b04a53f2b4f6
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio-port.c
@@ -0,0 +1,194 @@
+/*
+ * SiRF Audio port driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-port.h"
+
+struct sirf_audio_port {
+	struct regmap *regmap;
+	struct snd_dmaengine_dai_dma_data playback_dma_data;
+	struct snd_dmaengine_dai_dma_data capture_dma_data;
+};
+
+static void sirf_audio_port_tx_enable(struct sirf_audio_port *port)
+{
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+		AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+	regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0);
+	regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+		AUDIO_FIFO_START, AUDIO_FIFO_START);
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL,
+		IC_TX_ENABLE, IC_TX_ENABLE);
+}
+
+static void sirf_audio_port_tx_disable(struct sirf_audio_port *port)
+{
+	regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL,
+		IC_TX_ENABLE, ~IC_TX_ENABLE);
+}
+
+static void sirf_audio_port_rx_enable(struct sirf_audio_port *port,
+	int channels)
+{
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+		AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+	regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_INT_MSK, 0);
+	regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0);
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+		AUDIO_FIFO_START, AUDIO_FIFO_START);
+	if (channels == 1)
+		regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+			IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO);
+	else
+		regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+			IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO);
+}
+
+static void sirf_audio_port_rx_disable(struct sirf_audio_port *port)
+{
+	regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+			IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO);
+}
+
+static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai)
+{
+	struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_init_dma_data(dai, &port->playback_dma_data,
+			&port->capture_dma_data);
+	return 0;
+}
+
+static int sirf_audio_port_trigger(struct snd_pcm_substream *substream, int cmd,
+	struct snd_soc_dai *dai)
+{
+	struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai);
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (playback)
+			sirf_audio_port_tx_disable(port);
+		else
+			sirf_audio_port_rx_disable(port);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (playback)
+			sirf_audio_port_tx_enable(port);
+		else
+			sirf_audio_port_rx_enable(port,
+				substream->runtime->channels);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops sirf_audio_port_dai_ops = {
+	.trigger = sirf_audio_port_trigger,
+};
+
+static struct snd_soc_dai_driver sirf_audio_port_dai = {
+	.probe = sirf_audio_port_dai_probe,
+	.name = "sirf-audio-port",
+	.id = 0,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &sirf_audio_port_dai_ops,
+};
+
+static const struct snd_soc_component_driver sirf_audio_port_component = {
+	.name       = "sirf-audio-port",
+};
+
+static const struct regmap_config sirf_audio_port_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK,
+	.cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_port_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct sirf_audio_port *port;
+	void __iomem *base;
+	struct resource *mem_res;
+
+	port = devm_kzalloc(&pdev->dev,
+			sizeof(struct sirf_audio_port), GFP_KERNEL);
+	if (!port)
+		return -ENOMEM;
+
+	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem_res) {
+		dev_err(&pdev->dev, "no mem resource?\n");
+		return -ENODEV;
+	}
+
+	base = devm_ioremap(&pdev->dev, mem_res->start,
+			resource_size(mem_res));
+	if (base == NULL)
+		return -ENOMEM;
+
+	port->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+					    &sirf_audio_port_regmap_config);
+	if (IS_ERR(port->regmap))
+		return PTR_ERR(port->regmap);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+			&sirf_audio_port_component, &sirf_audio_port_dai, 1);
+	if (ret)
+		return ret;
+
+	platform_set_drvdata(pdev, port);
+	return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+}
+
+static const struct of_device_id sirf_audio_port_of_match[] = {
+	{ .compatible = "sirf,audio-port", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_port_of_match);
+
+static struct platform_driver sirf_audio_port_driver = {
+	.driver = {
+		.name = "sirf-audio-port",
+		.owner = THIS_MODULE,
+		.of_match_table = sirf_audio_port_of_match,
+	},
+	.probe = sirf_audio_port_probe,
+};
+
+module_platform_driver(sirf_audio_port_driver);
+
+MODULE_DESCRIPTION("SiRF Audio Port driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/sirf/sirf-audio-port.h b/sound/soc/sirf/sirf-audio-port.h
new file mode 100644
index 000000000000..f32dc54f4499
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio-port.h
@@ -0,0 +1,62 @@
+/*
+ * SiRF Audio port controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_PORT_H
+#define _SIRF_AUDIO_PORT_H
+
+#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK     0x3F
+#define AUDIO_PORT_TX_FIFO_SC_OFFSET    0
+#define AUDIO_PORT_TX_FIFO_LC_OFFSET    10
+#define AUDIO_PORT_TX_FIFO_HC_OFFSET    20
+
+#define TX_FIFO_SC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+				<< AUDIO_PORT_TX_FIFO_SC_OFFSET)
+#define TX_FIFO_LC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+				<< AUDIO_PORT_TX_FIFO_LC_OFFSET)
+#define TX_FIFO_HC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+				<< AUDIO_PORT_TX_FIFO_HC_OFFSET)
+
+#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK     0x0F
+#define AUDIO_PORT_RX_FIFO_SC_OFFSET    0
+#define AUDIO_PORT_RX_FIFO_LC_OFFSET    10
+#define AUDIO_PORT_RX_FIFO_HC_OFFSET    20
+
+#define RX_FIFO_SC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+				<< AUDIO_PORT_RX_FIFO_SC_OFFSET)
+#define RX_FIFO_LC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+				<< AUDIO_PORT_RX_FIFO_LC_OFFSET)
+#define RX_FIFO_HC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+				<< AUDIO_PORT_RX_FIFO_HC_OFFSET)
+#define AUDIO_PORT_IC_CODEC_TX_CTRL		(0x00F4)
+#define AUDIO_PORT_IC_CODEC_RX_CTRL		(0x00F8)
+
+#define AUDIO_PORT_IC_TXFIFO_OP			(0x00FC)
+#define AUDIO_PORT_IC_TXFIFO_LEV_CHK		(0x0100)
+#define AUDIO_PORT_IC_TXFIFO_STS		(0x0104)
+#define AUDIO_PORT_IC_TXFIFO_INT		(0x0108)
+#define AUDIO_PORT_IC_TXFIFO_INT_MSK		(0x010C)
+
+#define AUDIO_PORT_IC_RXFIFO_OP			(0x0110)
+#define AUDIO_PORT_IC_RXFIFO_LEV_CHK		(0x0114)
+#define AUDIO_PORT_IC_RXFIFO_STS		(0x0118)
+#define AUDIO_PORT_IC_RXFIFO_INT		(0x011C)
+#define AUDIO_PORT_IC_RXFIFO_INT_MSK		(0x0120)
+
+#define AUDIO_FIFO_START		(1 << 0)
+#define AUDIO_FIFO_RESET		(1 << 1)
+
+#define AUDIO_FIFO_FULL			(1 << 0)
+#define AUDIO_FIFO_EMPTY		(1 << 1)
+#define AUDIO_FIFO_OFLOW		(1 << 2)
+#define AUDIO_FIFO_UFLOW		(1 << 3)
+
+#define IC_TX_ENABLE		(0x03)
+#define IC_RX_ENABLE_MONO	(0x01)
+#define IC_RX_ENABLE_STEREO	(0x03)
+
+#endif /*__SIRF_AUDIO_PORT_H*/
-- 
cgit v1.2.3


From af12a31f054f55b75c8cf4a459c7bd9d1c7726a9 Mon Sep 17 00:00:00 2001
From: Rongjun Ying <rongjun.ying@csr.com>
Date: Wed, 5 Mar 2014 16:34:36 +0800
Subject: ASoC: sirf: Add SiRF audio card

This connects platform DAI, SiRF internal audio codec DAI  and
SiRF auido port DAI together and works as a mach driver.

Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/sirf-audio.txt       |  41 ++++++
 sound/soc/sirf/Kconfig                             |   6 +
 sound/soc/sirf/Makefile                            |   2 +
 sound/soc/sirf/sirf-audio.c                        | 156 +++++++++++++++++++++
 4 files changed, 205 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt
 create mode 100644 sound/soc/sirf/sirf-audio.c

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/sirf-audio.txt b/Documentation/devicetree/bindings/sound/sirf-audio.txt
new file mode 100644
index 000000000000..c88882ca3704
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio.txt
@@ -0,0 +1,41 @@
+* SiRF atlas6 and prima2 internal audio codec and port based audio setups
+
+Required properties:
+- compatible: "sirf,sirf-audio-card"
+- sirf,audio-platform: phandle for the platform node
+- sirf,audio-codec: phandle for the SiRF internal codec node
+
+Optional properties:
+- hp-pa-gpios: Need to be present if the board need control external
+  headphone amplifier.
+- spk-pa-gpios: Need to be present if the board need control external
+  speaker amplifier.
+- hp-switch-gpios: Need to be present if the board capable to detect jack
+  insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Ext Spk
+ * Line In
+ * Mic
+
+SiRF internal audio codec pins:
+ * HPOUTL
+ * HPOUTR
+ * SPKOUT
+ * Ext Mic
+ * Mic Bias
+
+Example:
+
+sound {
+		compatible = "sirf,sirf-audio-card";
+		sirf,audio-codec = <&audiocodec>;
+		sirf,audio-platform = <&audioport>;
+		hp-pa-gpios = <&gpio 44 0>;
+		spk-pa-gpios = <&gpio 46 0>;
+		hp-switch-gpios = <&gpio 45 0>;
+};
+
diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig
index 75b0344d2151..89e89429b04a 100644
--- a/sound/soc/sirf/Kconfig
+++ b/sound/soc/sirf/Kconfig
@@ -3,6 +3,12 @@ config SND_SOC_SIRF
 	depends on ARCH_SIRF || COMPILE_TEST
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
+config SND_SOC_SIRF_AUDIO
+	tristate "SoC Audio support for SiRF internal audio codec"
+	depends on SND_SOC_SIRF
+	select SND_SOC_SIRF_AUDIO_CODEC
+	select SND_SOC_SIRF_AUDIO_PORT
+
 config SND_SOC_SIRF_AUDIO_PORT
 	select REGMAP_MMIO
 	tristate
diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile
index fb012c852b28..913b93231d4e 100644
--- a/sound/soc/sirf/Makefile
+++ b/sound/soc/sirf/Makefile
@@ -1,3 +1,5 @@
+snd-soc-sirf-audio-objs := sirf-audio.o
 snd-soc-sirf-audio-port-objs := sirf-audio-port.o
 
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o
 obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o
diff --git a/sound/soc/sirf/sirf-audio.c b/sound/soc/sirf/sirf-audio.c
new file mode 100644
index 000000000000..ecef51021653
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio.c
@@ -0,0 +1,156 @@
+/*
+ * SiRF audio card driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+struct sirf_audio_card {
+	unsigned int            gpio_hp_pa;
+	unsigned int            gpio_spk_pa;
+};
+
+static int sirf_audio_hp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *ctrl, int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card);
+	int on = !SND_SOC_DAPM_EVENT_OFF(event);
+	if (gpio_is_valid(sirf_audio_card->gpio_hp_pa))
+		gpio_set_value(sirf_audio_card->gpio_hp_pa, on);
+	return 0;
+}
+
+static int sirf_audio_spk_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *ctrl, int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card);
+	int on = !SND_SOC_DAPM_EVENT_OFF(event);
+
+	if (gpio_is_valid(sirf_audio_card->gpio_spk_pa))
+		gpio_set_value(sirf_audio_card->gpio_spk_pa, on);
+
+	return 0;
+}
+static const struct snd_soc_dapm_widget sirf_audio_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Hp", sirf_audio_hp_event),
+	SND_SOC_DAPM_SPK("Ext Spk", sirf_audio_spk_event),
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"Hp", NULL, "HPOUTL"},
+	{"Hp", NULL, "HPOUTR"},
+	{"Ext Spk", NULL, "SPKOUT"},
+	{"MICIN1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Ext Mic"},
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sirf_audio_dai_link[] = {
+	{
+		.name = "SiRF audio card",
+		.stream_name = "SiRF audio HiFi",
+		.codec_dai_name = "sirf-audio-codec",
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sirf_audio_card = {
+	.name = "SiRF audio card",
+	.owner = THIS_MODULE,
+	.dai_link = sirf_audio_dai_link,
+	.num_links = ARRAY_SIZE(sirf_audio_dai_link),
+	.dapm_widgets = sirf_audio_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(sirf_audio_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
+};
+
+static int sirf_audio_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_sirf_audio_card;
+	struct sirf_audio_card *sirf_audio_card;
+	int ret;
+
+	sirf_audio_card = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_card),
+			GFP_KERNEL);
+	if (sirf_audio_card == NULL)
+		return -ENOMEM;
+
+	sirf_audio_dai_link[0].cpu_of_node =
+		of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0);
+	sirf_audio_dai_link[0].platform_of_node =
+		of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0);
+	sirf_audio_dai_link[0].codec_of_node =
+		of_parse_phandle(pdev->dev.of_node, "sirf,audio-codec", 0);
+	sirf_audio_card->gpio_spk_pa = of_get_named_gpio(pdev->dev.of_node,
+			"spk-pa-gpios", 0);
+	sirf_audio_card->gpio_hp_pa =  of_get_named_gpio(pdev->dev.of_node,
+			"hp-pa-gpios", 0);
+	if (gpio_is_valid(sirf_audio_card->gpio_spk_pa)) {
+		ret = devm_gpio_request_one(&pdev->dev,
+				sirf_audio_card->gpio_spk_pa,
+				GPIOF_OUT_INIT_LOW, "SPA_PA_SD");
+		if (ret) {
+			dev_err(&pdev->dev,
+				"Failed to request GPIO_%d for reset: %d\n",
+				sirf_audio_card->gpio_spk_pa, ret);
+			return ret;
+		}
+	}
+	if (gpio_is_valid(sirf_audio_card->gpio_hp_pa)) {
+		ret = devm_gpio_request_one(&pdev->dev,
+				sirf_audio_card->gpio_hp_pa,
+				GPIOF_OUT_INIT_LOW, "HP_PA_SD");
+		if (ret) {
+			dev_err(&pdev->dev,
+				"Failed to request GPIO_%d for reset: %d\n",
+				sirf_audio_card->gpio_hp_pa, ret);
+			return ret;
+		}
+	}
+
+	card->dev = &pdev->dev;
+	snd_soc_card_set_drvdata(card, sirf_audio_card);
+
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
+
+	return ret;
+}
+
+static const struct of_device_id sirf_audio_of_match[] = {
+	{.compatible = "sirf,sirf-audio-card", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_of_match);
+
+static struct platform_driver sirf_audio_driver = {
+	.driver = {
+		.name = "sirf-audio-card",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = sirf_audio_of_match,
+	},
+	.probe = sirf_audio_probe,
+};
+module_platform_driver(sirf_audio_driver);
+
+MODULE_AUTHOR("RongJun Ying <RongJun.Ying@csr.com>");
+MODULE_DESCRIPTION("ALSA SoC SIRF audio card driver");
+MODULE_LICENSE("GPL v2");
-- 
cgit v1.2.3


From e00447fafbf7daf2cd49205b97e63d9734068a4f Mon Sep 17 00:00:00 2001
From: Jyri Sarha <jsarha@ti.com>
Date: Tue, 11 Mar 2014 12:57:32 +0200
Subject: ASoC: tlv320aic31xx: Add basic codec driver implementation

This commit adds a bare bones driver support for TLV320AIC31XX family
audio codecs. The driver adds basic stereo playback trough headphone
and speaker outputs and mono capture trough microphone inputs.

The driver is currently missing support at least for mini DSP features
and jack detection. I have tested the driver only on TLV320AIC3111,
but based on the data sheets TLV320AIC3100, TLV320AIC3110, and
TLV320AIC3120 should work Ok too.

The base for the implementation was taken from:
git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx
-branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/tlv320aic31xx.txt    |   61 +
 include/dt-bindings/sound/tlv320aic31xx-micbias.h  |    8 +
 sound/soc/codecs/Kconfig                           |    4 +
 sound/soc/codecs/Makefile                          |    2 +
 sound/soc/codecs/tlv320aic31xx.c                   | 1295 ++++++++++++++++++++
 sound/soc/codecs/tlv320aic31xx.h                   |  258 ++++
 6 files changed, 1628 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
 create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h
 create mode 100644 sound/soc/codecs/tlv320aic31xx.c
 create mode 100644 sound/soc/codecs/tlv320aic31xx.h

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
new file mode 100644
index 000000000000..74c66dee3e14
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@@ -0,0 +1,61 @@
+Texas Instruments - tlv320aic31xx Codec module
+
+The tlv320aic31xx serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+    "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp
+    "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp
+    "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP)
+    "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
+    "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
+    "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
+
+- reg - <int> -  I2C slave address
+
+
+Optional properties:
+
+- gpio-reset - gpio pin number used for codec reset
+- ai31xx-micbias-vg - MicBias Voltage setting
+        1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
+        2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
+        3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD
+	If this node is not mentioned or if the value is unknown, then
+	micbias	is set to 2.0V.
+- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
+  DVDD-supply : power supplies for the device as covered in
+  Documentation/devicetree/bindings/regulator/regulator.txt
+
+CODEC output pins:
+  * HPL
+  * HPR
+  * SPL, devices with stereo speaker amp
+  * SPR, devices with stereo speaker amp
+  * SPK, devices with mono speaker amp
+  * MICBIAS
+
+CODEC input pins:
+  * MIC1LP
+  * MIC1RP
+  * MIC1LM
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+tlv320aic31xx: tlv320aic31xx@18 {
+	compatible = "ti,tlv320aic311x";
+	reg = <0x18>;
+
+	ai31xx-micbias-vg = <MICBIAS_OFF>;
+
+	HPVDD-supply = <&regulator>;
+	SPRVDD-supply = <&regulator>;
+	SPLVDD-supply = <&regulator>;
+	AVDD-supply = <&regulator>;
+	IOVDD-supply = <&regulator>;
+	DVDD-supply = <&regulator>;
+};
diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h
new file mode 100644
index 000000000000..f5cb772ab9c8
--- /dev/null
+++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h
@@ -0,0 +1,8 @@
+#ifndef __DT_TLV320AIC31XX_MICBIAS_H
+#define __DT_TLV320AIC31XX_MICBIAS_H
+
+#define MICBIAS_2_0V		1
+#define MICBIAS_2_5V		2
+#define MICBIAS_AVDDV		3
+
+#endif /* __DT_TLV320AIC31XX_MICBIAS_H */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 983d087aa92a..66f6c53ea328 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -73,6 +73,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TAS5086 if I2C
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
+	select SND_SOC_TLV320AIC31XX if I2C
 	select SND_SOC_TLV320AIC32X4 if I2C
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
@@ -361,6 +362,9 @@ config SND_SOC_TLV320AIC26
 	tristate
 	depends on SPI
 
+config SND_SOC_TLV320AIC31XX
+        tristate
+
 config SND_SOC_TLV320AIC32X4
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1deeb20fd411..ff1775c562fe 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -64,6 +64,7 @@ snd-soc-stac9766-objs := stac9766.o
 snd-soc-tas5086-objs := tas5086.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
+snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
 snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
@@ -194,6 +195,7 @@ obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TAS5086)	+= snd-soc-tas5086.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
+obj-$(CONFIG_SND_SOC_TLV320AIC31XX)     += snd-soc-tlv320aic31xx.o
 obj-$(CONFIG_SND_SOC_TLV320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
new file mode 100644
index 000000000000..e60e37b43a1b
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -0,0 +1,1295 @@
+/*
+ * ALSA SoC TLV320AIC31XX codec driver
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * Based on ground work by: Ajit Kulkarni <x0175765@ti.com>
+ *
+ * This package 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.
+ *
+ * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * The TLV320AIC31xx series of audio codec is a low-power, highly integrated
+ * high performance codec which provides a stereo DAC, a mono ADC,
+ * and mono/stereo Class-D speaker driver.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+#include "tlv320aic31xx.h"
+
+static const struct reg_default aic31xx_reg_defaults[] = {
+	{ AIC31XX_CLKMUX, 0x00 },
+	{ AIC31XX_PLLPR, 0x11 },
+	{ AIC31XX_PLLJ, 0x04 },
+	{ AIC31XX_PLLDMSB, 0x00 },
+	{ AIC31XX_PLLDLSB, 0x00 },
+	{ AIC31XX_NDAC, 0x01 },
+	{ AIC31XX_MDAC, 0x01 },
+	{ AIC31XX_DOSRMSB, 0x00 },
+	{ AIC31XX_DOSRLSB, 0x80 },
+	{ AIC31XX_NADC, 0x01 },
+	{ AIC31XX_MADC, 0x01 },
+	{ AIC31XX_AOSR, 0x80 },
+	{ AIC31XX_IFACE1, 0x00 },
+	{ AIC31XX_DATA_OFFSET, 0x00 },
+	{ AIC31XX_IFACE2, 0x00 },
+	{ AIC31XX_BCLKN, 0x01 },
+	{ AIC31XX_DACSETUP, 0x14 },
+	{ AIC31XX_DACMUTE, 0x0c },
+	{ AIC31XX_LDACVOL, 0x00 },
+	{ AIC31XX_RDACVOL, 0x00 },
+	{ AIC31XX_ADCSETUP, 0x00 },
+	{ AIC31XX_ADCFGA, 0x80 },
+	{ AIC31XX_ADCVOL, 0x00 },
+	{ AIC31XX_HPDRIVER, 0x04 },
+	{ AIC31XX_SPKAMP, 0x06 },
+	{ AIC31XX_DACMIXERROUTE, 0x00 },
+	{ AIC31XX_LANALOGHPL, 0x7f },
+	{ AIC31XX_RANALOGHPR, 0x7f },
+	{ AIC31XX_LANALOGSPL, 0x7f },
+	{ AIC31XX_RANALOGSPR, 0x7f },
+	{ AIC31XX_HPLGAIN, 0x02 },
+	{ AIC31XX_HPRGAIN, 0x02 },
+	{ AIC31XX_SPLGAIN, 0x00 },
+	{ AIC31XX_SPRGAIN, 0x00 },
+	{ AIC31XX_MICBIAS, 0x00 },
+	{ AIC31XX_MICPGA, 0x80 },
+	{ AIC31XX_MICPGAPI, 0x00 },
+	{ AIC31XX_MICPGAMI, 0x00 },
+};
+
+static bool aic31xx_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case AIC31XX_PAGECTL: /* regmap implementation requires this */
+	case AIC31XX_RESET: /* always clears after write */
+	case AIC31XX_OT_FLAG:
+	case AIC31XX_ADCFLAG:
+	case AIC31XX_DACFLAG1:
+	case AIC31XX_DACFLAG2:
+	case AIC31XX_OFFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG2:
+	case AIC31XX_INTRADCFLAG2:
+		return true;
+	}
+	return false;
+}
+
+static bool aic31xx_writeable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case AIC31XX_OT_FLAG:
+	case AIC31XX_ADCFLAG:
+	case AIC31XX_DACFLAG1:
+	case AIC31XX_DACFLAG2:
+	case AIC31XX_OFFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG2:
+	case AIC31XX_INTRADCFLAG2:
+		return false;
+	}
+	return true;
+}
+
+static const struct regmap_range_cfg aic31xx_ranges[] = {
+	{
+		.range_min = 0,
+		.range_max = 12 * 128,
+		.selector_reg = AIC31XX_PAGECTL,
+		.selector_mask = 0xff,
+		.selector_shift = 0,
+		.window_start = 0,
+		.window_len = 128,
+	},
+};
+
+struct regmap_config aic31xx_i2c_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.writeable_reg = aic31xx_writeable,
+	.volatile_reg = aic31xx_volatile,
+	.reg_defaults = aic31xx_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+	.ranges = aic31xx_ranges,
+	.num_ranges = ARRAY_SIZE(aic31xx_ranges),
+	.max_register = 12 * 128,
+};
+
+#define AIC31XX_NUM_SUPPLIES	6
+static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = {
+	"HPVDD",
+	"SPRVDD",
+	"SPLVDD",
+	"AVDD",
+	"IOVDD",
+	"DVDD",
+};
+
+struct aic31xx_disable_nb {
+	struct notifier_block nb;
+	struct aic31xx_priv *aic31xx;
+};
+
+struct aic31xx_priv {
+	struct snd_soc_codec *codec;
+	u8 i2c_regs_status;
+	struct device *dev;
+	struct regmap *regmap;
+	struct aic31xx_pdata pdata;
+	struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES];
+	struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES];
+	unsigned int sysclk;
+	int rate_div_line;
+};
+
+struct aic31xx_rate_divs {
+	u32 mclk;
+	u32 rate;
+	u8 p_val;
+	u8 pll_j;
+	u16 pll_d;
+	u16 dosr;
+	u8 ndac;
+	u8 mdac;
+	u8 aosr;
+	u8 nadc;
+	u8 madc;
+};
+
+/* ADC dividers can be disabled by cofiguring them to 0 */
+static const struct aic31xx_rate_divs aic31xx_divs[] = {
+	/* mclk      rate  pll: p  j	 d     dosr ndac mdac  aors nadc madc */
+	/* 8k rate */
+	{12000000,   8000,	1, 8, 1920,	128,  48,  2,	128,  48,  2},
+	{24000000,   8000,	2, 8, 1920,	128,  48,  2,	128,  48,  2},
+	{25000000,   8000,	2, 7, 8643,	128,  48,  2,	128,  48,  2},
+	/* 11.025k rate */
+	{12000000,  11025,	1, 7, 5264,	128,  32,  2,	128,  32,  2},
+	{24000000,  11025,	2, 7, 5264,	128,  32,  2,	128,  32,  2},
+	{25000000,  11025,	2, 7, 2253,	128,  32,  2,	128,  32,  2},
+	/* 16k rate */
+	{12000000,  16000,	1, 8, 1920,	128,  24,  2,	128,  24,  2},
+	{24000000,  16000,	2, 8, 1920,	128,  24,  2,	128,  24,  2},
+	{25000000,  16000,	2, 7, 8643,	128,  24,  2,	128,  24,  2},
+	/* 22.05k rate */
+	{12000000,  22050,	1, 7, 5264,	128,  16,  2,	128,  16,  2},
+	{24000000,  22050,	2, 7, 5264,	128,  16,  2,	128,  16,  2},
+	{25000000,  22050,	2, 7, 2253,	128,  16,  2,	128,  16,  2},
+	/* 32k rate */
+	{12000000,  32000,	1, 8, 1920,	128,  12,  2,	128,  12,  2},
+	{24000000,  32000,	2, 8, 1920,	128,  12,  2,	128,  12,  2},
+	{25000000,  32000,	2, 7, 8643,	128,  12,  2,	128,  12,  2},
+	/* 44.1k rate */
+	{12000000,  44100,	1, 7, 5264,	128,   8,  2,	128,   8,  2},
+	{24000000,  44100,	2, 7, 5264,	128,   8,  2,	128,   8,  2},
+	{25000000,  44100,	2, 7, 2253,	128,   8,  2,	128,   8,  2},
+	/* 48k rate */
+	{12000000,  48000,	1, 8, 1920,	128,   8,  2,	128,   8,  2},
+	{24000000,  48000,	2, 8, 1920,	128,   8,  2,	128,   8,  2},
+	{25000000,  48000,	2, 7, 8643,	128,   8,  2,	128,   8,  2},
+	/* 88.2k rate */
+	{12000000,  88200,	1, 7, 5264,	 64,   8,  2,	 64,   8,  2},
+	{24000000,  88200,	2, 7, 5264,	 64,   8,  2,	 64,   8,  2},
+	{25000000,  88200,	2, 7, 2253,	 64,   8,  2,	 64,   8,  2},
+	/* 96k rate */
+	{12000000,  96000,	1, 8, 1920,	 64,   8,  2,	 64,   8,  2},
+	{24000000,  96000,	2, 8, 1920,	 64,   8,  2,	 64,   8,  2},
+	{25000000,  96000,	2, 7, 8643,	 64,   8,  2,	 64,   8,  2},
+	/* 176.4k rate */
+	{12000000, 176400,	1, 7, 5264,	 32,   8,  2,	 32,   8,  2},
+	{24000000, 176400,	2, 7, 5264,	 32,   8,  2,	 32,   8,  2},
+	{25000000, 176400,	2, 7, 2253,	 32,   8,  2,	 32,   8,  2},
+	/* 192k rate */
+	{12000000, 192000,	1, 8, 1920,	 32,   8,  2,	 32,   8,  2},
+	{24000000, 192000,	2, 8, 1920,	 32,   8,  2,	 32,   8,  2},
+	{25000000, 192000,	2, 7, 8643,	 32,   8,  2,	 32,   8,  2},
+};
+
+static const char * const ldac_in_text[] = {
+	"Off", "Left Data", "Right Data", "Mono"
+};
+
+static const char * const rdac_in_text[] = {
+	"Off", "Right Data", "Left Data", "Mono"
+};
+
+static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text);
+
+static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text);
+
+static const char * const mic_select_text[] = {
+	"Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm"
+};
+
+static const
+SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text);
+
+static const
+SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text);
+
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0);
+static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
+
+/*
+ * controls to be exported to the user space
+ */
+static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
+	SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
+			   AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),
+
+	SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
+		       adc_fgain_tlv),
+
+	SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1),
+	SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL,
+			   0, -24, 40, 6, 0, adc_cgain_tlv),
+
+	SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
+		       119, 0, mic_pga_tlv),
+
+	SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
+		     AIC31XX_HPRGAIN, 2, 1, 0),
+	SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
+			 AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
+
+	SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
+			 AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic311x_snd_controls[] = {
+	SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
+		     AIC31XX_SPRGAIN, 2, 1, 0),
+	SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
+			 AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv),
+
+	SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
+			 AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic310x_snd_controls[] = {
+	SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
+		   2, 1, 0),
+	SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
+		       3, 3, 0, class_D_drv_tlv),
+
+	SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
+		       0, 0x7F, 1, sp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new ldac_in_control =
+	SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum);
+
+static const struct snd_kcontrol_new rdac_in_control =
+	SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum);
+
+int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
+		      unsigned int mask, unsigned int wbits, int sleep,
+		      int count)
+{
+	unsigned int bits;
+	int counter = count;
+	int ret = regmap_read(aic31xx->regmap, reg, &bits);
+	while ((bits & mask) != wbits && counter && !ret) {
+		usleep_range(sleep, sleep * 2);
+		ret = regmap_read(aic31xx->regmap, reg, &bits);
+		counter--;
+	}
+	if ((bits & mask) != wbits) {
+		dev_err(aic31xx->dev,
+			"%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n",
+			__func__, reg, bits, wbits, ret, mask,
+			(count - counter) * sleep);
+		ret = -1;
+	}
+	return ret;
+}
+
+#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg))
+
+static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
+				    struct snd_kcontrol *kcontrol, int event)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec);
+	unsigned int reg = AIC31XX_DACFLAG1;
+	unsigned int mask;
+
+	switch (WIDGET_BIT(w->reg, w->shift)) {
+	case WIDGET_BIT(AIC31XX_DACSETUP, 7):
+		mask = AIC31XX_LDACPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_DACSETUP, 6):
+		mask = AIC31XX_RDACPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_HPDRIVER, 7):
+		mask = AIC31XX_HPLDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_HPDRIVER, 6):
+		mask = AIC31XX_HPRDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_SPKAMP, 7):
+		mask = AIC31XX_SPLDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_SPKAMP, 6):
+		mask = AIC31XX_SPRDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_ADCSETUP, 7):
+		mask = AIC31XX_ADCPWRSTATUS_MASK;
+		reg = AIC31XX_ADCFLAG;
+		break;
+	default:
+		dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
+			w->name, __func__);
+		return -EINVAL;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);
+	case SND_SOC_DAPM_POST_PMD:
+		return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
+	default:
+		dev_dbg(w->codec->dev,
+			"Unhandled dapm widget event %d from %s\n",
+			event, w->name);
+	}
+	return 0;
+}
+
+static const struct snd_kcontrol_new left_output_switches[] = {
+	SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
+	SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
+	SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_output_switches[] = {
+	SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
+	SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new p_term_mic1lp =
+	SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);
+
+static const struct snd_kcontrol_new p_term_mic1rp =
+	SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum);
+
+static const struct snd_kcontrol_new p_term_mic1lm =
+	SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum);
+
+static const struct snd_kcontrol_new m_term_mic1lm =
+	SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum);
+
+static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_spl_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_spr_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0);
+
+static int mic_bias_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* change mic bias voltage to user defined */
+		snd_soc_update_bits(codec, AIC31XX_MICBIAS,
+				    AIC31XX_MICBIAS_MASK,
+				    aic31xx->pdata.micbias_vg <<
+				    AIC31XX_MICBIAS_SHIFT);
+		dev_dbg(codec->dev, "%s: turned on\n", __func__);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		/* turn mic bias off */
+		snd_soc_update_bits(codec, AIC31XX_MICBIAS,
+				    AIC31XX_MICBIAS_MASK, 0);
+		dev_dbg(codec->dev, "%s: turned off\n", __func__);
+		break;
+	}
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_MUX("DAC Left Input",
+			 SND_SOC_NOPM, 0, 0, &ldac_in_control),
+	SND_SOC_DAPM_MUX("DAC Right Input",
+			 SND_SOC_NOPM, 0, 0, &rdac_in_control),
+	/* DACs */
+	SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback",
+			   AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback",
+			   AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+			   left_output_switches,
+			   ARRAY_SIZE(left_output_switches)),
+	SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+			   right_output_switches,
+			   ARRAY_SIZE(right_output_switches)),
+
+	SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_hpl_switch),
+	SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_hpr_switch),
+
+	/* Output drivers */
+	SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0,
+			       NULL, 0, aic31xx_dapm_power_event,
+			       SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0,
+			       NULL, 0, aic31xx_dapm_power_event,
+			       SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
+
+	/* ADC */
+	SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
+			   aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			   SND_SOC_DAPM_POST_PMD),
+
+	/* Input Selection to MIC_PGA */
+	SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
+			 &p_term_mic1lp),
+	SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0,
+			 &p_term_mic1rp),
+	SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
+			 &p_term_mic1lm),
+
+	SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
+			 &m_term_mic1lm),
+	/* Enabling & Disabling MIC Gain Ctl */
+	SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
+			 7, 1, NULL, 0),
+
+	/* Mic Bias */
+	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
+			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MIC1LP"),
+	SND_SOC_DAPM_INPUT("MIC1RP"),
+	SND_SOC_DAPM_INPUT("MIC1LM"),
+};
+
+static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
+	/* AIC3111 and AIC3110 have stereo class-D amplifier */
+	SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
+			       aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0,
+			       aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_spl_switch),
+	SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_spr_switch),
+	SND_SOC_DAPM_OUTPUT("SPL"),
+	SND_SOC_DAPM_OUTPUT("SPR"),
+};
+
+/* AIC3100 and AIC3120 have only mono class-D amplifier */
+static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
+			       aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_spl_switch),
+	SND_SOC_DAPM_OUTPUT("SPK"),
+};
+
+static const struct snd_soc_dapm_route
+aic31xx_audio_map[] = {
+	/* DAC Input Routing */
+	{"DAC Left Input", "Left Data", "DAC IN"},
+	{"DAC Left Input", "Right Data", "DAC IN"},
+	{"DAC Left Input", "Mono", "DAC IN"},
+	{"DAC Right Input", "Left Data", "DAC IN"},
+	{"DAC Right Input", "Right Data", "DAC IN"},
+	{"DAC Right Input", "Mono", "DAC IN"},
+	{"DAC Left", NULL, "DAC Left Input"},
+	{"DAC Right", NULL, "DAC Right Input"},
+
+	/* Mic input */
+	{"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
+	{"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
+	{"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"},
+	{"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"},
+	{"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"},
+	{"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"},
+	{"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"},
+	{"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"},
+	{"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"},
+
+	{"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"},
+	{"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"},
+	{"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"},
+
+	{"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"},
+	{"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"},
+	{"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"},
+	{"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"},
+
+	{"ADC", NULL, "MIC_GAIN_CTL"},
+
+	/* Left Output */
+	{"Output Left", "From Left DAC", "DAC Left"},
+	{"Output Left", "From MIC1LP", "MIC1LP"},
+	{"Output Left", "From MIC1RP", "MIC1RP"},
+
+	/* Right Output */
+	{"Output Right", "From Right DAC", "DAC Right"},
+	{"Output Right", "From MIC1RP", "MIC1RP"},
+
+	/* HPL path */
+	{"HP Left", "Switch", "Output Left"},
+	{"HPL Driver", NULL, "HP Left"},
+	{"HPL", NULL, "HPL Driver"},
+
+	/* HPR path */
+	{"HP Right", "Switch", "Output Right"},
+	{"HPR Driver", NULL, "HP Right"},
+	{"HPR", NULL, "HPR Driver"},
+};
+
+static const struct snd_soc_dapm_route
+aic311x_audio_map[] = {
+	/* SP L path */
+	{"Speaker Left", "Switch", "Output Left"},
+	{"SPL ClassD", NULL, "Speaker Left"},
+	{"SPL", NULL, "SPL ClassD"},
+
+	/* SP R path */
+	{"Speaker Right", "Switch", "Output Right"},
+	{"SPR ClassD", NULL, "Speaker Right"},
+	{"SPR", NULL, "SPR ClassD"},
+};
+
+static const struct snd_soc_dapm_route
+aic310x_audio_map[] = {
+	/* SP L path */
+	{"Speaker", "Switch", "Output Left"},
+	{"SPK ClassD", NULL, "Speaker"},
+	{"SPK", NULL, "SPK ClassD"},
+};
+
+static int aic31xx_add_controls(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+
+	if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT)
+		ret = snd_soc_add_codec_controls(
+			codec, aic311x_snd_controls,
+			ARRAY_SIZE(aic311x_snd_controls));
+	else
+		ret = snd_soc_add_codec_controls(
+			codec, aic310x_snd_controls,
+			ARRAY_SIZE(aic310x_snd_controls));
+
+	return ret;
+}
+
+static int aic31xx_add_widgets(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
+		ret = snd_soc_dapm_new_controls(
+			dapm, aic311x_dapm_widgets,
+			ARRAY_SIZE(aic311x_dapm_widgets));
+		if (ret)
+			return ret;
+
+		ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map,
+					      ARRAY_SIZE(aic311x_audio_map));
+		if (ret)
+			return ret;
+	} else {
+		ret = snd_soc_dapm_new_controls(
+			dapm, aic310x_dapm_widgets,
+			ARRAY_SIZE(aic310x_dapm_widgets));
+		if (ret)
+			return ret;
+
+		ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map,
+					      ARRAY_SIZE(aic310x_audio_map));
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int aic31xx_setup_pll(struct snd_soc_codec *codec,
+			     struct snd_pcm_hw_params *params)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int bclk_n = 0;
+	int i;
+
+	/* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */
+	snd_soc_update_bits(codec, AIC31XX_CLKMUX,
+			    AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL);
+	snd_soc_update_bits(codec, AIC31XX_IFACE2,
+			    AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK);
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
+		if (aic31xx_divs[i].rate == params_rate(params) &&
+		    aic31xx_divs[i].mclk == aic31xx->sysclk)
+			break;
+	}
+
+	if (i == ARRAY_SIZE(aic31xx_divs)) {
+		dev_err(codec->dev, "%s: Sampling rate %u not supported\n",
+			__func__, params_rate(params));
+		return -EINVAL;
+	}
+
+	/* PLL configuration */
+	snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
+			    (aic31xx_divs[i].p_val << 4) | 0x01);
+	snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j);
+
+	snd_soc_write(codec, AIC31XX_PLLDMSB,
+		      aic31xx_divs[i].pll_d >> 8);
+	snd_soc_write(codec, AIC31XX_PLLDLSB,
+		      aic31xx_divs[i].pll_d & 0xff);
+
+	/* DAC dividers configuration */
+	snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].ndac);
+	snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].mdac);
+
+	snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8);
+	snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff);
+
+	/* ADC dividers configuration. Write reset value 1 if not used. */
+	snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1);
+	snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1);
+
+	snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr);
+
+	/* Bit clock divider configuration. */
+	bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac)
+		/ snd_soc_params_to_frame_size(params);
+	if (bclk_n == 0) {
+		dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AIC31XX_BCLKN,
+			    AIC31XX_PLL_MASK, bclk_n);
+
+	aic31xx->rate_div_line = i;
+
+	dev_dbg(codec->dev,
+		"pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n",
+		aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d,
+		aic31xx_divs[i].p_val, aic31xx_divs[i].dosr,
+		aic31xx_divs[i].ndac, aic31xx_divs[i].mdac,
+		aic31xx_divs[i].aosr, aic31xx_divs[i].nadc,
+		aic31xx_divs[i].madc, bclk_n);
+
+	return 0;
+}
+
+static int aic31xx_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *tmp)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	u8 data = 0;
+
+	dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n",
+		__func__, params_format(params), params_width(params),
+		params_rate(params));
+
+	switch (params_width(params)) {
+	case 16:
+		break;
+	case 20:
+		data = (AIC31XX_WORD_LEN_20BITS <<
+			AIC31XX_IFACE1_DATALEN_SHIFT);
+		break;
+	case 24:
+		data = (AIC31XX_WORD_LEN_24BITS <<
+			AIC31XX_IFACE1_DATALEN_SHIFT);
+		break;
+	case 32:
+		data = (AIC31XX_WORD_LEN_32BITS <<
+			AIC31XX_IFACE1_DATALEN_SHIFT);
+		break;
+	default:
+		dev_err(codec->dev, "%s: Unsupported format %d\n",
+			__func__, params_format(params));
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AIC31XX_IFACE1,
+			    AIC31XX_IFACE1_DATALEN_MASK,
+			    data);
+
+	return aic31xx_setup_pll(codec, params);
+}
+
+static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	if (mute) {
+		snd_soc_update_bits(codec, AIC31XX_DACMUTE,
+				    AIC31XX_DACMUTE_MASK,
+				    AIC31XX_DACMUTE_MASK);
+	} else {
+		snd_soc_update_bits(codec, AIC31XX_DACMUTE,
+				    AIC31XX_DACMUTE_MASK, 0x0);
+	}
+
+	return 0;
+}
+
+static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 iface_reg1 = 0;
+	u8 iface_reg3 = 0;
+	u8 dsp_a_val = 0;
+
+	dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER;
+		break;
+	default:
+		dev_alert(codec->dev, "Invalid DAI master/slave interface\n");
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		dsp_a_val = 0x1;
+	case SND_SOC_DAIFMT_DSP_B:
+		/* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			iface_reg3 |= AIC31XX_BCLKINV_MASK;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			break;
+		default:
+			return -EINVAL;
+		}
+		iface_reg1 |= (AIC31XX_DSP_MODE <<
+			       AIC31XX_IFACE1_DATATYPE_SHIFT);
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE <<
+			       AIC31XX_IFACE1_DATATYPE_SHIFT);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE <<
+			       AIC31XX_IFACE1_DATATYPE_SHIFT);
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI interface format\n");
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AIC31XX_IFACE1,
+			    AIC31XX_IFACE1_DATATYPE_MASK |
+			    AIC31XX_IFACE1_MASTER_MASK,
+			    iface_reg1);
+	snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET,
+			    AIC31XX_DATA_OFFSET_MASK,
+			    dsp_a_val);
+	snd_soc_update_bits(codec, AIC31XX_IFACE2,
+			    AIC31XX_BCLKINV_MASK,
+			    iface_reg3);
+
+	return 0;
+}
+
+static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n",
+		__func__, clk_id, freq, dir);
+
+	for (i = 0; aic31xx_divs[i].mclk != freq; i++) {
+		if (i == ARRAY_SIZE(aic31xx_divs)) {
+			dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
+				__func__, freq);
+			return -EINVAL;
+		}
+	}
+
+	/* set clock on MCLK, BCLK, or GPIO1 as PLL input */
+	snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK,
+			    clk_id << AIC31XX_PLL_CLKIN_SHIFT);
+
+	aic31xx->sysclk = freq;
+	return 0;
+}
+
+static int aic31xx_regulator_event(struct notifier_block *nb,
+				   unsigned long event, void *data)
+{
+	struct aic31xx_disable_nb *disable_nb =
+		container_of(nb, struct aic31xx_disable_nb, nb);
+	struct aic31xx_priv *aic31xx = disable_nb->aic31xx;
+
+	if (event & REGULATOR_EVENT_DISABLE) {
+		/*
+		 * Put codec to reset and as at least one of the
+		 * supplies was disabled.
+		 */
+		if (gpio_is_valid(aic31xx->pdata.gpio_reset))
+			gpio_set_value(aic31xx->pdata.gpio_reset, 0);
+
+		regcache_mark_dirty(aic31xx->regmap);
+		dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__);
+	}
+
+	return 0;
+}
+
+static void aic31xx_clk_on(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	u8 mask = AIC31XX_PM_MASK;
+	u8 on = AIC31XX_PM_MASK;
+
+	dev_dbg(codec->dev, "codec clock -> on (rate %d)\n",
+		aic31xx_divs[aic31xx->rate_div_line].rate);
+	snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on);
+	mdelay(10);
+	snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on);
+	snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on);
+	if (aic31xx_divs[aic31xx->rate_div_line].nadc)
+		snd_soc_update_bits(codec, AIC31XX_NADC, mask, on);
+	if (aic31xx_divs[aic31xx->rate_div_line].madc)
+		snd_soc_update_bits(codec, AIC31XX_MADC, mask, on);
+	snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on);
+}
+
+static void aic31xx_clk_off(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	u8 mask = AIC31XX_PM_MASK;
+	u8 off = 0;
+
+	dev_dbg(codec->dev, "codec clock -> off\n");
+	snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_MADC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_NADC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off);
+}
+
+static int aic31xx_power_on(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies),
+				    aic31xx->supplies);
+	if (ret)
+		return ret;
+
+	if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {
+		gpio_set_value(aic31xx->pdata.gpio_reset, 1);
+		udelay(100);
+	}
+	regcache_cache_only(aic31xx->regmap, false);
+	ret = regcache_sync(aic31xx->regmap);
+	if (ret != 0) {
+		dev_err(codec->dev,
+			"Failed to restore cache: %d\n", ret);
+		regcache_cache_only(aic31xx->regmap, true);
+		regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
+				       aic31xx->supplies);
+		return ret;
+	}
+	return 0;
+}
+
+static int aic31xx_power_off(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	regcache_cache_only(aic31xx->regmap, true);
+	ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
+				     aic31xx->supplies);
+
+	return ret;
+}
+
+static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
+		codec->dapm.bias_level, level);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+			aic31xx_clk_on(codec);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		switch (codec->dapm.bias_level) {
+		case SND_SOC_BIAS_OFF:
+			aic31xx_power_on(codec);
+			break;
+		case SND_SOC_BIAS_PREPARE:
+			aic31xx_clk_off(codec);
+			break;
+		default:
+			BUG();
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		aic31xx_power_off(codec);
+		break;
+	}
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static int aic31xx_suspend(struct snd_soc_codec *codec)
+{
+	aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int aic31xx_resume(struct snd_soc_codec *codec)
+{
+	aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int aic31xx_codec_probe(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	dev_dbg(aic31xx->dev, "## %s\n", __func__);
+
+	aic31xx = snd_soc_codec_get_drvdata(codec);
+	codec->control_data = aic31xx->regmap;
+
+	aic31xx->codec = codec;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+
+	if (ret != 0) {
+		dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n",
+			ret);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) {
+		aic31xx->disable_nb[i].nb.notifier_call =
+			aic31xx_regulator_event;
+		aic31xx->disable_nb[i].aic31xx = aic31xx;
+		ret = regulator_register_notifier(aic31xx->supplies[i].consumer,
+						  &aic31xx->disable_nb[i].nb);
+		if (ret) {
+			dev_err(codec->dev,
+				"Failed to request regulator notifier: %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	regcache_cache_only(aic31xx->regmap, true);
+	regcache_mark_dirty(aic31xx->regmap);
+
+	ret = aic31xx_add_controls(codec);
+	if (ret)
+		return ret;
+
+	ret = aic31xx_add_widgets(codec);
+
+	return ret;
+}
+
+static int aic31xx_codec_remove(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int i;
+	/* power down chip */
+	aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
+		regulator_unregister_notifier(aic31xx->supplies[i].consumer,
+					      &aic31xx->disable_nb[i].nb);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
+	.probe			= aic31xx_codec_probe,
+	.remove			= aic31xx_codec_remove,
+	.suspend		= aic31xx_suspend,
+	.resume			= aic31xx_resume,
+	.set_bias_level		= aic31xx_set_bias_level,
+	.controls		= aic31xx_snd_controls,
+	.num_controls		= ARRAY_SIZE(aic31xx_snd_controls),
+	.dapm_widgets		= aic31xx_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(aic31xx_dapm_widgets),
+	.dapm_routes		= aic31xx_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(aic31xx_audio_map),
+};
+
+static struct snd_soc_dai_ops aic31xx_dai_ops = {
+	.hw_params	= aic31xx_hw_params,
+	.set_sysclk	= aic31xx_set_dai_sysclk,
+	.set_fmt	= aic31xx_set_dai_fmt,
+	.digital_mute	= aic31xx_dac_mute,
+};
+
+static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
+	{
+		.name = "tlv320aic31xx-hifi",
+		.playback = {
+			.stream_name	 = "Playback",
+			.channels_min	 = 1,
+			.channels_max	 = 2,
+			.rates		 = AIC31XX_RATES,
+			.formats	 = AIC31XX_FORMATS,
+		},
+		.capture = {
+			.stream_name	 = "Capture",
+			.channels_min	 = 1,
+			.channels_max	 = 2,
+			.rates		 = AIC31XX_RATES,
+			.formats	 = AIC31XX_FORMATS,
+		},
+		.ops = &aic31xx_dai_ops,
+		.symmetric_rates = 1,
+	}
+};
+
+#if defined(CONFIG_OF)
+static const struct of_device_id tlv320aic31xx_of_match[] = {
+	{ .compatible = "ti,tlv320aic310x" },
+	{ .compatible = "ti,tlv320aic311x" },
+	{ .compatible = "ti,tlv320aic3100" },
+	{ .compatible = "ti,tlv320aic3110" },
+	{ .compatible = "ti,tlv320aic3120" },
+	{ .compatible = "ti,tlv320aic3111" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match);
+
+static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
+{
+	struct device_node *np = aic31xx->dev->of_node;
+	unsigned int value = MICBIAS_2_0V;
+	int ret;
+
+	of_property_read_u32(np, "ai31xx-micbias-vg", &value);
+	switch (value) {
+	case MICBIAS_2_0V:
+	case MICBIAS_2_5V:
+	case MICBIAS_AVDDV:
+		aic31xx->pdata.micbias_vg = value;
+		break;
+	default:
+		dev_err(aic31xx->dev,
+			"Bad ai31xx-micbias-vg value %d DT\n",
+			value);
+		aic31xx->pdata.micbias_vg = MICBIAS_2_0V;
+	}
+
+	ret = of_get_named_gpio(np, "gpio-reset", 0);
+	if (ret > 0)
+		aic31xx->pdata.gpio_reset = ret;
+}
+#else /* CONFIG_OF */
+static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
+{
+}
+#endif /* CONFIG_OF */
+
+void aic31xx_device_init(struct aic31xx_priv *aic31xx)
+{
+	int ret, i;
+
+	dev_set_drvdata(aic31xx->dev, aic31xx);
+
+	if (dev_get_platdata(aic31xx->dev))
+		memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev),
+		       sizeof(aic31xx->pdata));
+	else if (aic31xx->dev->of_node)
+		aic31xx_pdata_from_of(aic31xx);
+
+	if (aic31xx->pdata.gpio_reset) {
+		ret = devm_gpio_request_one(aic31xx->dev,
+					    aic31xx->pdata.gpio_reset,
+					    GPIOF_OUT_INIT_HIGH,
+					    "aic31xx-reset-pin");
+		if (ret < 0) {
+			dev_err(aic31xx->dev, "not able to acquire gpio\n");
+			return;
+		}
+	}
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
+		aic31xx->supplies[i].supply = aic31xx_supply_names[i];
+
+	ret = devm_regulator_bulk_get(aic31xx->dev,
+				      ARRAY_SIZE(aic31xx->supplies),
+				      aic31xx->supplies);
+	if (ret != 0)
+		dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
+
+}
+
+static int aic31xx_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct aic31xx_priv *aic31xx;
+	int ret;
+	const struct regmap_config *regmap_config;
+
+	dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__,
+		id->name, (int) id->driver_data);
+
+	regmap_config = &aic31xx_i2c_regmap;
+
+	aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL);
+	if (aic31xx == NULL)
+		return -ENOMEM;
+
+	aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config);
+
+	if (IS_ERR(aic31xx->regmap)) {
+		ret = PTR_ERR(aic31xx->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+	aic31xx->dev = &i2c->dev;
+
+	aic31xx->pdata.codec_type = id->driver_data;
+
+	aic31xx_device_init(aic31xx);
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
+				     aic31xx_dai_driver,
+				     ARRAY_SIZE(aic31xx_dai_driver));
+
+	return ret;
+}
+
+static int aic31xx_i2c_remove(struct i2c_client *i2c)
+{
+	struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev);
+
+	kfree(aic31xx);
+	return 0;
+}
+
+static const struct i2c_device_id aic31xx_i2c_id[] = {
+	{ "tlv320aic310x", AIC3100 },
+	{ "tlv320aic311x", AIC3110 },
+	{ "tlv320aic3100", AIC3100 },
+	{ "tlv320aic3110", AIC3110 },
+	{ "tlv320aic3120", AIC3120 },
+	{ "tlv320aic3111", AIC3111 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
+
+static struct i2c_driver aic31xx_i2c_driver = {
+	.driver = {
+		.name	= "tlv320aic31xx-codec",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(tlv320aic31xx_of_match),
+	},
+	.probe		= aic31xx_i2c_probe,
+	.remove		= (aic31xx_i2c_remove),
+	.id_table	= aic31xx_i2c_id,
+};
+
+module_i2c_driver(aic31xx_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver");
+MODULE_AUTHOR("Jyri Sarha");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
new file mode 100644
index 000000000000..52ed57c69dfa
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -0,0 +1,258 @@
+/*
+ * ALSA SoC TLV320AIC31XX codec driver
+ *
+ * Copyright (C) 2013 Texas Instruments, Inc.
+ *
+ * This package 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.
+ *
+ * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ */
+#ifndef _TLV320AIC31XX_H
+#define _TLV320AIC31XX_H
+
+#define AIC31XX_RATES	SNDRV_PCM_RATE_8000_192000
+
+#define AIC31XX_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+
+#define AIC31XX_STEREO_CLASS_D_BIT	0x1
+#define AIC31XX_MINIDSP_BIT		0x2
+
+enum aic31xx_type {
+	AIC3100	= 0,
+	AIC3110 = AIC31XX_STEREO_CLASS_D_BIT,
+	AIC3120 = AIC31XX_MINIDSP_BIT,
+	AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT),
+};
+
+struct aic31xx_pdata {
+	enum aic31xx_type codec_type;
+	unsigned int gpio_reset;
+	int micbias_vg;
+};
+
+/* Page Control Register */
+#define AIC31XX_PAGECTL				0x00
+
+/* Page 0 Registers */
+/* Software reset register */
+#define AIC31XX_RESET				0x01
+/* OT FLAG register */
+#define AIC31XX_OT_FLAG				0x03
+/* Clock clock Gen muxing, Multiplexers*/
+#define AIC31XX_CLKMUX				0x04
+/* PLL P and R-VAL register */
+#define AIC31XX_PLLPR				0x05
+/* PLL J-VAL register */
+#define AIC31XX_PLLJ				0x06
+/* PLL D-VAL MSB register */
+#define AIC31XX_PLLDMSB				0x07
+/* PLL D-VAL LSB register */
+#define AIC31XX_PLLDLSB				0x08
+/* DAC NDAC_VAL register*/
+#define AIC31XX_NDAC				0x0B
+/* DAC MDAC_VAL register */
+#define AIC31XX_MDAC				0x0C
+/* DAC OSR setting register 1, MSB value */
+#define AIC31XX_DOSRMSB				0x0D
+/* DAC OSR setting register 2, LSB value */
+#define AIC31XX_DOSRLSB				0x0E
+#define AIC31XX_MINI_DSP_INPOL			0x10
+/* Clock setting register 8, PLL */
+#define AIC31XX_NADC				0x12
+/* Clock setting register 9, PLL */
+#define AIC31XX_MADC				0x13
+/* ADC Oversampling (AOSR) Register */
+#define AIC31XX_AOSR				0x14
+/* Clock setting register 9, Multiplexers */
+#define AIC31XX_CLKOUTMUX			0x19
+/* Clock setting register 10, CLOCKOUT M divider value */
+#define AIC31XX_CLKOUTMVAL			0x1A
+/* Audio Interface Setting Register 1 */
+#define AIC31XX_IFACE1				0x1B
+/* Audio Data Slot Offset Programming */
+#define AIC31XX_DATA_OFFSET			0x1C
+/* Audio Interface Setting Register 2 */
+#define AIC31XX_IFACE2				0x1D
+/* Clock setting register 11, BCLK N Divider */
+#define AIC31XX_BCLKN				0x1E
+/* Audio Interface Setting Register 3, Secondary Audio Interface */
+#define AIC31XX_IFACESEC1			0x1F
+/* Audio Interface Setting Register 4 */
+#define AIC31XX_IFACESEC2			0x20
+/* Audio Interface Setting Register 5 */
+#define AIC31XX_IFACESEC3			0x21
+/* I2C Bus Condition */
+#define AIC31XX_I2C				0x22
+/* ADC FLAG */
+#define AIC31XX_ADCFLAG				0x24
+/* DAC Flag Registers */
+#define AIC31XX_DACFLAG1			0x25
+#define AIC31XX_DACFLAG2			0x26
+/* Sticky Interrupt flag (overflow) */
+#define AIC31XX_OFFLAG				0x27
+/* Sticy DAC Interrupt flags */
+#define AIC31XX_INTRDACFLAG			0x2C
+/* Sticy ADC Interrupt flags */
+#define AIC31XX_INTRADCFLAG			0x2D
+/* DAC Interrupt flags 2 */
+#define AIC31XX_INTRDACFLAG2			0x2E
+/* ADC Interrupt flags 2 */
+#define AIC31XX_INTRADCFLAG2			0x2F
+/* INT1 interrupt control */
+#define AIC31XX_INT1CTRL			0x30
+/* INT2 interrupt control */
+#define AIC31XX_INT2CTRL			0x31
+/* GPIO1 control */
+#define AIC31XX_GPIO1				0x33
+
+#define AIC31XX_DACPRB				0x3C
+/* ADC Instruction Set Register */
+#define AIC31XX_ADCPRB				0x3D
+/* DAC channel setup register */
+#define AIC31XX_DACSETUP			0x3F
+/* DAC Mute and volume control register */
+#define AIC31XX_DACMUTE				0x40
+/* Left DAC channel digital volume control */
+#define AIC31XX_LDACVOL				0x41
+/* Right DAC channel digital volume control */
+#define AIC31XX_RDACVOL				0x42
+/* Headset detection */
+#define AIC31XX_HSDETECT			0x43
+/* ADC Digital Mic */
+#define AIC31XX_ADCSETUP			0x51
+/* ADC Digital Volume Control Fine Adjust */
+#define AIC31XX_ADCFGA				0x52
+/* ADC Digital Volume Control Coarse Adjust */
+#define AIC31XX_ADCVOL				0x53
+
+
+/* Page 1 Registers */
+/* Headphone drivers */
+#define AIC31XX_HPDRIVER			0x9F
+/* Class-D Speakear Amplifier */
+#define AIC31XX_SPKAMP				0xA0
+/* HP Output Drivers POP Removal Settings */
+#define AIC31XX_HPPOP				0xA1
+/* Output Driver PGA Ramp-Down Period Control */
+#define AIC31XX_SPPGARAMP			0xA2
+/* DAC_L and DAC_R Output Mixer Routing */
+#define AIC31XX_DACMIXERROUTE			0xA3
+/* Left Analog Vol to HPL */
+#define AIC31XX_LANALOGHPL			0xA4
+/* Right Analog Vol to HPR */
+#define AIC31XX_RANALOGHPR			0xA5
+/* Left Analog Vol to SPL */
+#define AIC31XX_LANALOGSPL			0xA6
+/* Right Analog Vol to SPR */
+#define AIC31XX_RANALOGSPR			0xA7
+/* HPL Driver */
+#define AIC31XX_HPLGAIN				0xA8
+/* HPR Driver */
+#define AIC31XX_HPRGAIN				0xA9
+/* SPL Driver */
+#define AIC31XX_SPLGAIN				0xAA
+/* SPR Driver */
+#define AIC31XX_SPRGAIN				0xAB
+/* HP Driver Control */
+#define AIC31XX_HPCONTROL			0xAC
+/* MIC Bias Control */
+#define AIC31XX_MICBIAS				0xAE
+/* MIC PGA*/
+#define AIC31XX_MICPGA				0xAF
+/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */
+#define AIC31XX_MICPGAPI			0xB0
+/* ADC Input Selection for M-Terminal */
+#define AIC31XX_MICPGAMI			0xB1
+/* Input CM Settings */
+#define AIC31XX_MICPGACM			0xB2
+
+/* Bits, masks and shifts */
+
+/* AIC31XX_CLKMUX */
+#define AIC31XX_PLL_CLKIN_MASK			0x0c
+#define AIC31XX_PLL_CLKIN_SHIFT			2
+#define AIC31XX_PLL_CLKIN_MCLK			0
+#define AIC31XX_CODEC_CLKIN_MASK		0x03
+#define AIC31XX_CODEC_CLKIN_SHIFT		0
+#define AIC31XX_CODEC_CLKIN_PLL			3
+#define AIC31XX_CODEC_CLKIN_BCLK		1
+
+/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC,
+   AIC31XX_BCLKN */
+#define AIC31XX_PLL_MASK		0x7f
+#define AIC31XX_PM_MASK			0x80
+
+/* AIC31XX_IFACE1 */
+#define AIC31XX_WORD_LEN_16BITS		0x00
+#define AIC31XX_WORD_LEN_20BITS		0x01
+#define AIC31XX_WORD_LEN_24BITS		0x02
+#define AIC31XX_WORD_LEN_32BITS		0x03
+#define AIC31XX_IFACE1_DATALEN_MASK	0x30
+#define AIC31XX_IFACE1_DATALEN_SHIFT	(4)
+#define AIC31XX_IFACE1_DATATYPE_MASK	0xC0
+#define AIC31XX_IFACE1_DATATYPE_SHIFT	(6)
+#define AIC31XX_I2S_MODE		0x00
+#define AIC31XX_DSP_MODE		0x01
+#define AIC31XX_RIGHT_JUSTIFIED_MODE	0x02
+#define AIC31XX_LEFT_JUSTIFIED_MODE	0x03
+#define AIC31XX_IFACE1_MASTER_MASK	0x0C
+#define AIC31XX_BCLK_MASTER		0x08
+#define AIC31XX_WCLK_MASTER		0x04
+
+/* AIC31XX_DATA_OFFSET */
+#define AIC31XX_DATA_OFFSET_MASK	0xFF
+
+/* AIC31XX_IFACE2 */
+#define AIC31XX_BCLKINV_MASK		0x08
+#define AIC31XX_BDIVCLK_MASK		0x03
+#define AIC31XX_DAC2BCLK		0x00
+#define AIC31XX_DACMOD2BCLK		0x01
+#define AIC31XX_ADC2BCLK		0x02
+#define AIC31XX_ADCMOD2BCLK		0x03
+
+/* AIC31XX_ADCFLAG */
+#define AIC31XX_ADCPWRSTATUS_MASK		0x40
+
+/* AIC31XX_DACFLAG1 */
+#define AIC31XX_LDACPWRSTATUS_MASK		0x80
+#define AIC31XX_RDACPWRSTATUS_MASK		0x08
+#define AIC31XX_HPLDRVPWRSTATUS_MASK		0x20
+#define AIC31XX_HPRDRVPWRSTATUS_MASK		0x02
+#define AIC31XX_SPLDRVPWRSTATUS_MASK		0x10
+#define AIC31XX_SPRDRVPWRSTATUS_MASK		0x01
+
+/* AIC31XX_INTRDACFLAG */
+#define AIC31XX_HPSCDETECT_MASK			0x80
+#define AIC31XX_BUTTONPRESS_MASK		0x20
+#define AIC31XX_HSPLUG_MASK			0x10
+#define AIC31XX_LDRCTHRES_MASK			0x08
+#define AIC31XX_RDRCTHRES_MASK			0x04
+#define AIC31XX_DACSINT_MASK			0x02
+#define AIC31XX_DACAINT_MASK			0x01
+
+/* AIC31XX_INT1CTRL */
+#define AIC31XX_HSPLUGDET_MASK			0x80
+#define AIC31XX_BUTTONPRESSDET_MASK		0x40
+#define AIC31XX_DRCTHRES_MASK			0x20
+#define AIC31XX_AGCNOISE_MASK			0x10
+#define AIC31XX_OC_MASK				0x08
+#define AIC31XX_ENGINE_MASK			0x04
+
+/* AIC31XX_DACSETUP */
+#define AIC31XX_SOFTSTEP_MASK			0x03
+
+/* AIC31XX_DACMUTE */
+#define AIC31XX_DACMUTE_MASK			0x0C
+
+/* AIC31XX_MICBIAS */
+#define AIC31XX_MICBIAS_MASK			0x03
+#define AIC31XX_MICBIAS_SHIFT			0
+
+#endif	/* _TLV320AIC31XX_H */
-- 
cgit v1.2.3


From cece5656901f09db13fbb569ff04f627ec2e0ab6 Mon Sep 17 00:00:00 2001
From: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Date: Wed, 12 Feb 2014 18:20:57 +0100
Subject: ASoC: add S/PDIF support to Armada 370 DB ASoC driver

The Armada 370 DB board not only has analog audio input/output, but
also S/PDIF input/output. This commit adds support for S/PDIF in the
ASoC machine driver of the Armada 370 DB platform, and adjusts the
Device Tree bindings documentation accordingly.

Signed-off-by: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../bindings/sound/armada-370db-audio.txt          |  9 ++++---
 sound/soc/kirkwood/Kconfig                         |  1 +
 sound/soc/kirkwood/armada-370-db.c                 | 28 ++++++++++++++++++++++
 3 files changed, 35 insertions(+), 3 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
index 3893b4d15a20..bf984d238620 100644
--- a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
+++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
@@ -11,14 +11,17 @@ Mandatory properties:
  * marvell,audio-controller: a phandle that points to the audio
    controller of the Armada 370 SoC.
 
- * marvell,audio-codec: a phandle that points to the analog audio
-   codec connected to the Armada 370 SoC.
+ * marvell,audio-codec: a set of three phandles that points to:
+
+    1/ the analog audio codec connected to the Armada 370 SoC
+    2/ the S/PDIF transceiver
+    3/ the S/PDIF receiver
 
 Example:
 
 	sound {
 	      compatible = "marvell,a370db-audio";
 	      marvell,audio-controller = <&audio_controller>;
-	      marvell,audio-codec = <&audio_codec>;
+	      marvell,audio-codec = <&audio_codec &spdif_out &spdif_in>;
 	      status = "okay";
 	};
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 2dc3ecf34801..49f8437665de 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -10,6 +10,7 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB
 	tristate "SoC Audio support for Armada 370 DB"
 	depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C
 	select SND_SOC_CS42L51
+	select SND_SOC_SPDIF
 	help
 	  Say Y if you want to add support for SoC audio on
 	  the Armada 370 Development Board.
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
index 977639b3ffde..c44333849259 100644
--- a/sound/soc/kirkwood/armada-370-db.c
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -67,6 +67,20 @@ static struct snd_soc_dai_link a370db_dai[] = {
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
 	.ops = &a370db_ops,
 },
+{
+	.name = "S/PDIF out",
+	.stream_name = "spdif-out",
+	.cpu_dai_name = "spdif",
+	.codec_dai_name = "dit-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+},
+{
+	.name = "S/PDIF in",
+	.stream_name = "spdif-in",
+	.cpu_dai_name = "spdif",
+	.codec_dai_name = "dir-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+},
 };
 
 static struct snd_soc_card a370db = {
@@ -95,6 +109,20 @@ static int a370db_probe(struct platform_device *pdev)
 		of_parse_phandle(pdev->dev.of_node,
 				 "marvell,audio-codec", 0);
 
+	a370db_dai[1].cpu_of_node = a370db_dai[0].cpu_of_node;
+	a370db_dai[1].platform_of_node = a370db_dai[0].cpu_of_node;
+
+	a370db_dai[1].codec_of_node =
+		of_parse_phandle(pdev->dev.of_node,
+				 "marvell,audio-codec", 1);
+
+	a370db_dai[2].cpu_of_node = a370db_dai[0].cpu_of_node;
+	a370db_dai[2].platform_of_node = a370db_dai[0].cpu_of_node;
+
+	a370db_dai[2].codec_of_node =
+		of_parse_phandle(pdev->dev.of_node,
+				 "marvell,audio-codec", 2);
+
 	return devm_snd_soc_register_card(card->dev, card);
 }
 
-- 
cgit v1.2.3


From 46c39cae292fd691f32e573e6c2c854e36614c93 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Wed, 12 Mar 2014 11:02:11 +0800
Subject: ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt

The current simple-card driver separates the daimft for cpu_dai and codec_dai.
So we might get different values for them (0x4003 and 0x1003 for example):

asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000
asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000
asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok

This is not allowed at all as we need to keep the DAIFMT settings identical
for both the ends of the link.

Thus this patch fixes it by overwriting the cpu_dai->fmt with codec_dai->fmt
since we defined the DAIFMT_MASTER basing on CODEC at the first place while
the other bits are same.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/simple-card.txt        |  6 ++++++
 sound/soc/generic/simple-card.c                      | 20 ++++++++++++++------
 2 files changed, 20 insertions(+), 6 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index b30c222f9cd3..881914b139ca 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -43,6 +43,12 @@ Optional CPU/CODEC subnodes properties:
 					  clock node (= common clock), or "system-clock-frequency"
 					  (if system doens't support common clock)
 
+Note:
+ * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and
+   'frame-inversion', the simple card will use the settings of CODEC for both
+   CPU and CODEC sides as we need to keep the settings identical for both ends
+   of the link.
+
 Example:
 
 sound {
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index ca7e63ef858a..2ee8ed56bcf1 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -151,6 +151,8 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 				     struct device *dev)
 {
 	struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
+	struct asoc_simple_dai *codec_dai = &priv->codec_dai;
+	struct asoc_simple_dai *cpu_dai = &priv->cpu_dai;
 	struct device_node *np;
 	char *name;
 	unsigned int daifmt;
@@ -184,7 +186,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	np = of_get_child_by_name(node, "simple-audio-card,cpu");
 	if (np) {
 		ret = asoc_simple_card_sub_parse_of(np, daifmt,
-						  &priv->cpu_dai,
+						  cpu_dai,
 						  &dai_link->cpu_of_node,
 						  &dai_link->cpu_dai_name);
 		of_node_put(np);
@@ -197,7 +199,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	np = of_get_child_by_name(node, "simple-audio-card,codec");
 	if (np) {
 		ret = asoc_simple_card_sub_parse_of(np, daifmt,
-						  &priv->codec_dai,
+						  codec_dai,
 						  &dai_link->codec_of_node,
 						  &dai_link->codec_dai_name);
 		of_node_put(np);
@@ -205,6 +207,12 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	if (ret < 0)
 		return ret;
 
+	/*
+	 * overwrite cpu_dai->fmt as its DAIFMT_MASTER bit is based on CODEC
+	 * while the other bits should be identical unless buggy SW/HW design.
+	 */
+	cpu_dai->fmt = codec_dai->fmt;
+
 	if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name)
 		return -EINVAL;
 
@@ -226,12 +234,12 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	dev_dbg(dev, "platform : %04x\n", daifmt);
 	dev_dbg(dev, "cpu : %s / %04x / %d\n",
 		dai_link->cpu_dai_name,
-		priv->cpu_dai.fmt,
-		priv->cpu_dai.sysclk);
+		cpu_dai->fmt,
+		cpu_dai->sysclk);
 	dev_dbg(dev, "codec : %s / %04x / %d\n",
 		dai_link->codec_dai_name,
-		priv->codec_dai.fmt,
-		priv->codec_dai.sysclk);
+		codec_dai->fmt,
+		codec_dai->sysclk);
 
 	/*
 	 * soc_bind_dai_link() will check cpu name
-- 
cgit v1.2.3


From 0c516b4ff85c0be4cee5b30ae59c9565c7f91a00 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Thu, 20 Mar 2014 18:18:37 +0800
Subject: ASoC: cs42xx8: Add codec driver support for CS42448/CS42888

This patch adds support for the Cirrus Logic CS42448/CS42888 Audio CODEC that
has six/four 24-bit AD and eight 24-bit DA converters.

[ CS42448/CS42888 supports both I2C and SPI control ports. As initial patch,
  this patch only adds the support for I2C. ]

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/cs42xx8.txt          |  28 +
 sound/soc/codecs/Kconfig                           |  10 +
 sound/soc/codecs/Makefile                          |   4 +
 sound/soc/codecs/cs42xx8-i2c.c                     |  64 +++
 sound/soc/codecs/cs42xx8.c                         | 602 +++++++++++++++++++++
 sound/soc/codecs/cs42xx8.h                         | 238 ++++++++
 6 files changed, 946 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt
 create mode 100644 sound/soc/codecs/cs42xx8-i2c.c
 create mode 100644 sound/soc/codecs/cs42xx8.c
 create mode 100644 sound/soc/codecs/cs42xx8.h

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt
new file mode 100644
index 000000000000..f631fbca6284
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt
@@ -0,0 +1,28 @@
+CS42448/CS42888 audio CODEC
+
+Required properties:
+
+  - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888"
+
+  - reg : the I2C address of the device for I2C
+
+  - clocks : a list of phandles + clock-specifiers, one for each entry in
+    clock-names
+
+  - clock-names : must contain "mclk"
+
+  - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device,
+    as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Example:
+
+codec: cs42888@48 {
+	compatible = "cirrus,cs42888";
+	reg = <0x48>;
+	clocks = <&codec_mclk 0>;
+	clock-names = "mclk";
+	VA-supply = <&reg_audio>;
+	VD-supply = <&reg_audio>;
+	VLS-supply = <&reg_audio>;
+	VLC-supply = <&reg_audio>;
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 983d087aa92a..a79c0d141f90 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS42L73 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_CS42XX8_I2C if I2C
 	select SND_SOC_CX20442 if TTY
 	select SND_SOC_DA7210 if I2C
 	select SND_SOC_DA7213 if I2C
@@ -254,6 +255,15 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_CS4271
 	tristate
 
+config SND_SOC_CS42XX8
+	tristate
+
+config SND_SOC_CS42XX8_I2C
+	tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)"
+	depends on I2C
+	select SND_SOC_CS42XX8
+	select REGMAP_I2C
+
 config SND_SOC_CX20442
 	tristate
 	depends on TTY
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bc126764a44d..cfe5d634c812 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,6 +23,8 @@ snd-soc-cs42l52-objs := cs42l52.o
 snd-soc-cs42l73-objs := cs42l73.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cs4271-objs := cs4271.o
+snd-soc-cs42xx8-objs := cs42xx8.o
+snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
 snd-soc-da7213-objs := da7213.o
@@ -156,6 +158,8 @@ obj-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
 obj-$(CONFIG_SND_SOC_CS42L73)	+= snd-soc-cs42l73.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
+obj-$(CONFIG_SND_SOC_CS42XX8)	+= snd-soc-cs42xx8.o
+obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DA7213)	+= snd-soc-da7213.o
diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c
new file mode 100644
index 000000000000..657dce27eade
--- /dev/null
+++ b/sound/soc/codecs/cs42xx8-i2c.c
@@ -0,0 +1,64 @@
+/*
+ * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.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/i2c.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/soc.h>
+
+#include "cs42xx8.h"
+
+static int cs42xx8_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	u32 ret = cs42xx8_probe(&i2c->dev,
+			devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config));
+	if (ret)
+		return ret;
+
+	pm_runtime_enable(&i2c->dev);
+	pm_request_idle(&i2c->dev);
+
+	return 0;
+}
+
+static int cs42xx8_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_codec(&i2c->dev);
+	pm_runtime_disable(&i2c->dev);
+
+	return 0;
+}
+
+static struct i2c_device_id cs42xx8_i2c_id[] = {
+	{"cs42448", (kernel_ulong_t)&cs42448_data},
+	{"cs42888", (kernel_ulong_t)&cs42888_data},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id);
+
+static struct i2c_driver cs42xx8_i2c_driver = {
+	.driver = {
+		.name = "cs42xx8",
+		.owner = THIS_MODULE,
+		.pm = &cs42xx8_pm,
+	},
+	.probe = cs42xx8_i2c_probe,
+	.remove = cs42xx8_i2c_remove,
+	.id_table = cs42xx8_i2c_id,
+};
+
+module_i2c_driver(cs42xx8_i2c_driver);
+
+MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver");
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
new file mode 100644
index 000000000000..082299a4e2fa
--- /dev/null
+++ b/sound/soc/codecs/cs42xx8.c
@@ -0,0 +1,602 @@
+/*
+ * Cirrus Logic CS42448/CS42888 Audio CODEC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.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/delay.h>
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "cs42xx8.h"
+
+#define CS42XX8_NUM_SUPPLIES 4
+static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = {
+	"VA",
+	"VD",
+	"VLS",
+	"VLC",
+};
+
+#define CS42XX8_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+			 SNDRV_PCM_FMTBIT_S20_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_LE | \
+			 SNDRV_PCM_FMTBIT_S32_LE)
+
+/* codec private data */
+struct cs42xx8_priv {
+	struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES];
+	const struct cs42xx8_driver_data *drvdata;
+	struct regmap *regmap;
+	struct clk *clk;
+
+	bool slave_mode;
+	unsigned long sysclk;
+};
+
+/* -127.5dB to 0dB with step of 0.5dB */
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+/* -64dB to 24dB with step of 0.5dB */
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0);
+
+static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" };
+static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross",
+					"Soft Ramp", "Soft Ramp on Zero Cross" };
+
+static const struct soc_enum adc1_single_enum =
+	SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single);
+static const struct soc_enum adc2_single_enum =
+	SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single);
+static const struct soc_enum adc3_single_enum =
+	SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single);
+static const struct soc_enum dac_szc_enum =
+	SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc);
+static const struct soc_enum adc_szc_enum =
+	SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc);
+
+static const struct snd_kcontrol_new cs42xx8_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1,
+			 CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3,
+			 CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv),
+	SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5,
+			 CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv),
+	SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7,
+			 CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv),
+	SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1,
+			   CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv),
+	SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3,
+			   CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv),
+	SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0),
+	SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0),
+	SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0),
+	SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0),
+	SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0),
+	SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0),
+	SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1),
+	SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0),
+	SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum),
+	SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum),
+	SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0),
+	SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum),
+	SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0),
+	SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0),
+	SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0),
+	SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum),
+};
+
+static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = {
+	SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5,
+			   CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv),
+	SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0),
+	SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum),
+};
+
+static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1),
+	SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1),
+	SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1),
+	SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1),
+
+	SND_SOC_DAPM_OUTPUT("AOUT1L"),
+	SND_SOC_DAPM_OUTPUT("AOUT1R"),
+	SND_SOC_DAPM_OUTPUT("AOUT2L"),
+	SND_SOC_DAPM_OUTPUT("AOUT2R"),
+	SND_SOC_DAPM_OUTPUT("AOUT3L"),
+	SND_SOC_DAPM_OUTPUT("AOUT3R"),
+	SND_SOC_DAPM_OUTPUT("AOUT4L"),
+	SND_SOC_DAPM_OUTPUT("AOUT4R"),
+
+	SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1),
+	SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1),
+
+	SND_SOC_DAPM_INPUT("AIN1L"),
+	SND_SOC_DAPM_INPUT("AIN1R"),
+	SND_SOC_DAPM_INPUT("AIN2L"),
+	SND_SOC_DAPM_INPUT("AIN2R"),
+
+	SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = {
+	SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1),
+
+	SND_SOC_DAPM_INPUT("AIN3L"),
+	SND_SOC_DAPM_INPUT("AIN3R"),
+};
+
+static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = {
+	/* Playback */
+	{ "AOUT1L", NULL, "DAC1" },
+	{ "AOUT1R", NULL, "DAC1" },
+	{ "DAC1", NULL, "PWR" },
+
+	{ "AOUT2L", NULL, "DAC2" },
+	{ "AOUT2R", NULL, "DAC2" },
+	{ "DAC2", NULL, "PWR" },
+
+	{ "AOUT3L", NULL, "DAC3" },
+	{ "AOUT3R", NULL, "DAC3" },
+	{ "DAC3", NULL, "PWR" },
+
+	{ "AOUT4L", NULL, "DAC4" },
+	{ "AOUT4R", NULL, "DAC4" },
+	{ "DAC4", NULL, "PWR" },
+
+	/* Capture */
+	{ "ADC1", NULL, "AIN1L" },
+	{ "ADC1", NULL, "AIN1R" },
+	{ "ADC1", NULL, "PWR" },
+
+	{ "ADC2", NULL, "AIN2L" },
+	{ "ADC2", NULL, "AIN2R" },
+	{ "ADC2", NULL, "PWR" },
+};
+
+static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = {
+	/* Capture */
+	{ "ADC3", NULL, "AIN3L" },
+	{ "ADC3", NULL, "AIN3R" },
+	{ "ADC3", NULL, "PWR" },
+};
+
+struct cs42xx8_ratios {
+	unsigned int ratio;
+	unsigned char speed;
+	unsigned char mclk;
+};
+
+static const struct cs42xx8_ratios cs42xx8_ratios[] = {
+	{ 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) },
+	{ 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) },
+	{ 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) },
+	{ 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) },
+	{ 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) },
+	{ 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) },
+	{ 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) },
+	{ 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) },
+	{ 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) }
+};
+
+static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+
+	cs42xx8->sysclk = freq;
+
+	return 0;
+}
+
+static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+	u32 val;
+
+	/* Set DAI format */
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_LEFT_J:
+		val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported dai format\n");
+		return -EINVAL;
+	}
+
+	regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF,
+			   CS42XX8_INTF_DAC_DIF_MASK |
+			   CS42XX8_INTF_ADC_DIF_MASK, val);
+
+	/* Set master/slave audio interface */
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs42xx8->slave_mode = true;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs42xx8->slave_mode = false;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported master/slave mode\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	u32 ratio = cs42xx8->sysclk / params_rate(params);
+	u32 i, fm, val, mask;
+
+	for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) {
+		if (cs42xx8_ratios[i].ratio == ratio)
+			break;
+	}
+
+	if (i == ARRAY_SIZE(cs42xx8_ratios)) {
+		dev_err(codec->dev, "unsupported sysclk ratio\n");
+		return -EINVAL;
+	}
+
+	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
+	val = cs42xx8_ratios[i].mclk;
+
+	fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed;
+
+	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
+			   CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask,
+			   CS42XX8_FUNCMOD_xC_FM(tx, fm) | val);
+
+	return 0;
+}
+
+static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE,
+			   CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops cs42xx8_dai_ops = {
+	.set_fmt	= cs42xx8_set_dai_fmt,
+	.set_sysclk	= cs42xx8_set_dai_sysclk,
+	.hw_params	= cs42xx8_hw_params,
+	.digital_mute	= cs42xx8_digital_mute,
+};
+
+static struct snd_soc_dai_driver cs42xx8_dai = {
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = CS42XX8_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = CS42XX8_FORMATS,
+	},
+	.ops = &cs42xx8_dai_ops,
+};
+
+static const struct reg_default cs42xx8_reg[] = {
+	{ 0x01, 0x01 },   /* Chip I.D. and Revision Register */
+	{ 0x02, 0x00 },   /* Power Control */
+	{ 0x03, 0xF0 },   /* Functional Mode */
+	{ 0x04, 0x46 },   /* Interface Formats */
+	{ 0x05, 0x00 },   /* ADC Control & DAC De-Emphasis */
+	{ 0x06, 0x10 },   /* Transition Control */
+	{ 0x07, 0x00 },   /* DAC Channel Mute */
+	{ 0x08, 0x00 },   /* Volume Control AOUT1 */
+	{ 0x09, 0x00 },   /* Volume Control AOUT2 */
+	{ 0x0a, 0x00 },   /* Volume Control AOUT3 */
+	{ 0x0b, 0x00 },   /* Volume Control AOUT4 */
+	{ 0x0c, 0x00 },   /* Volume Control AOUT5 */
+	{ 0x0d, 0x00 },   /* Volume Control AOUT6 */
+	{ 0x0e, 0x00 },   /* Volume Control AOUT7 */
+	{ 0x0f, 0x00 },   /* Volume Control AOUT8 */
+	{ 0x10, 0x00 },   /* DAC Channel Invert */
+	{ 0x11, 0x00 },   /* Volume Control AIN1 */
+	{ 0x12, 0x00 },   /* Volume Control AIN2 */
+	{ 0x13, 0x00 },   /* Volume Control AIN3 */
+	{ 0x14, 0x00 },   /* Volume Control AIN4 */
+	{ 0x15, 0x00 },   /* Volume Control AIN5 */
+	{ 0x16, 0x00 },   /* Volume Control AIN6 */
+	{ 0x17, 0x00 },   /* ADC Channel Invert */
+	{ 0x18, 0x00 },   /* Status Control */
+	{ 0x1a, 0x00 },   /* Status Mask */
+	{ 0x1b, 0x00 },   /* MUTEC Pin Control */
+};
+
+static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS42XX8_STATUS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS42XX8_CHIPID:
+	case CS42XX8_STATUS:
+		return false;
+	default:
+		return true;
+	}
+}
+
+const struct regmap_config cs42xx8_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = CS42XX8_LASTREG,
+	.reg_defaults = cs42xx8_reg,
+	.num_reg_defaults = ARRAY_SIZE(cs42xx8_reg),
+	.volatile_reg = cs42xx8_volatile_register,
+	.writeable_reg = cs42xx8_writeable_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(cs42xx8_regmap_config);
+
+static int cs42xx8_codec_probe(struct snd_soc_codec *codec)
+{
+	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	switch (cs42xx8->drvdata->num_adcs) {
+	case 3:
+		snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls,
+					ARRAY_SIZE(cs42xx8_adc3_snd_controls));
+		snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets,
+					ARRAY_SIZE(cs42xx8_adc3_dapm_widgets));
+		snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes,
+					ARRAY_SIZE(cs42xx8_adc3_dapm_routes));
+		break;
+	default:
+		break;
+	}
+
+	/* Mute all DAC channels */
+	regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL);
+
+	return 0;
+}
+
+static const struct snd_soc_codec_driver cs42xx8_driver = {
+	.probe = cs42xx8_codec_probe,
+	.idle_bias_off = true,
+
+	.controls = cs42xx8_snd_controls,
+	.num_controls = ARRAY_SIZE(cs42xx8_snd_controls),
+	.dapm_widgets = cs42xx8_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets),
+	.dapm_routes = cs42xx8_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes),
+};
+
+const struct cs42xx8_driver_data cs42448_data = {
+	.name = "cs42448",
+	.num_adcs = 3,
+};
+EXPORT_SYMBOL_GPL(cs42448_data);
+
+const struct cs42xx8_driver_data cs42888_data = {
+	.name = "cs42888",
+	.num_adcs = 2,
+};
+EXPORT_SYMBOL_GPL(cs42888_data);
+
+const struct of_device_id cs42xx8_of_match[] = {
+	{ .compatible = "cirrus,cs42448", .data = &cs42448_data, },
+	{ .compatible = "cirrus,cs42888", .data = &cs42888_data, },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, cs42xx8_of_match);
+EXPORT_SYMBOL_GPL(cs42xx8_of_match);
+
+int cs42xx8_probe(struct device *dev, struct regmap *regmap)
+{
+	const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev);
+	struct cs42xx8_priv *cs42xx8;
+	int ret, val, i;
+
+	cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL);
+	if (cs42xx8 == NULL)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, cs42xx8);
+
+	if (of_id)
+		cs42xx8->drvdata = of_id->data;
+
+	if (!cs42xx8->drvdata) {
+		dev_err(dev, "failed to find driver data\n");
+		return -EINVAL;
+	}
+
+	cs42xx8->clk = devm_clk_get(dev, "mclk");
+	if (IS_ERR(cs42xx8->clk)) {
+		dev_err(dev, "failed to get the clock: %ld\n",
+				PTR_ERR(cs42xx8->clk));
+		return -EINVAL;
+	}
+
+	cs42xx8->sysclk = clk_get_rate(cs42xx8->clk);
+
+	for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++)
+		cs42xx8->supplies[i].supply = cs42xx8_supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev,
+			ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies);
+	if (ret) {
+		dev_err(dev, "failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies),
+				    cs42xx8->supplies);
+	if (ret) {
+		dev_err(dev, "failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	/* Make sure hardware reset done */
+	msleep(5);
+
+	cs42xx8->regmap = regmap;
+	if (IS_ERR(cs42xx8->regmap)) {
+		ret = PTR_ERR(cs42xx8->regmap);
+		dev_err(dev, "failed to allocate regmap: %d\n", ret);
+		goto err_enable;
+	}
+
+	/*
+	 * We haven't marked the chip revision as volatile due to
+	 * sharing a register with the right input volume; explicitly
+	 * bypass the cache to read it.
+	 */
+	regcache_cache_bypass(cs42xx8->regmap, true);
+
+	/* Validate the chip ID */
+	regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val);
+	if (val < 0) {
+		dev_err(dev, "failed to get device ID: %x", val);
+		ret = -EINVAL;
+		goto err_enable;
+	}
+
+	/* The top four bits of the chip ID should be 0000 */
+	if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) {
+		dev_err(dev, "unmatched chip ID: %d\n",
+				val & CS42XX8_CHIPID_CHIP_ID_MASK);
+		ret = -EINVAL;
+		goto err_enable;
+	}
+
+	dev_info(dev, "found device, revision %X\n",
+			val & CS42XX8_CHIPID_REV_ID_MASK);
+
+	regcache_cache_bypass(cs42xx8->regmap, false);
+
+	cs42xx8_dai.name = cs42xx8->drvdata->name;
+
+	/* Each adc supports stereo input */
+	cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2;
+
+	ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1);
+	if (ret) {
+		dev_err(dev, "failed to register codec:%d\n", ret);
+		goto err_enable;
+	}
+
+	regcache_cache_only(cs42xx8->regmap, true);
+
+err_enable:
+	regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
+			       cs42xx8->supplies);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(cs42xx8_probe);
+
+#ifdef CONFIG_PM_RUNTIME
+static int cs42xx8_runtime_resume(struct device *dev)
+{
+	struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_prepare_enable(cs42xx8->clk);
+	if (ret) {
+		dev_err(dev, "failed to enable mclk: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies),
+				    cs42xx8->supplies);
+	if (ret) {
+		dev_err(dev, "failed to enable supplies: %d\n", ret);
+		goto err_clk;
+	}
+
+	/* Make sure hardware reset done */
+	msleep(5);
+
+	regcache_cache_only(cs42xx8->regmap, false);
+
+	ret = regcache_sync(cs42xx8->regmap);
+	if (ret) {
+		dev_err(dev, "failed to sync regmap: %d\n", ret);
+		goto err_bulk;
+	}
+
+	return 0;
+
+err_bulk:
+	regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
+			       cs42xx8->supplies);
+err_clk:
+	clk_disable_unprepare(cs42xx8->clk);
+
+	return ret;
+}
+
+static int cs42xx8_runtime_suspend(struct device *dev)
+{
+	struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev);
+
+	regcache_cache_only(cs42xx8->regmap, true);
+
+	regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
+			       cs42xx8->supplies);
+
+	clk_disable_unprepare(cs42xx8->clk);
+
+	return 0;
+}
+#endif
+
+const struct dev_pm_ops cs42xx8_pm = {
+	SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL)
+};
+EXPORT_SYMBOL_GPL(cs42xx8_pm);
+
+MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h
new file mode 100644
index 000000000000..da0b94aee419
--- /dev/null
+++ b/sound/soc/codecs/cs42xx8.h
@@ -0,0 +1,238 @@
+/*
+ * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver header file
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.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.
+ */
+
+#ifndef _CS42XX8_H
+#define _CS42XX8_H
+
+struct cs42xx8_driver_data {
+	char name[32];
+	int num_adcs;
+};
+
+extern const struct dev_pm_ops cs42xx8_pm;
+extern const struct cs42xx8_driver_data cs42448_data;
+extern const struct cs42xx8_driver_data cs42888_data;
+extern const struct regmap_config cs42xx8_regmap_config;
+int cs42xx8_probe(struct device *dev, struct regmap *regmap);
+
+/* CS42888 register map */
+#define CS42XX8_CHIPID				0x01	/* Chip ID */
+#define CS42XX8_PWRCTL				0x02	/* Power Control */
+#define CS42XX8_FUNCMOD				0x03	/* Functional Mode */
+#define CS42XX8_INTF				0x04	/* Interface Formats */
+#define CS42XX8_ADCCTL				0x05	/* ADC Control */
+#define CS42XX8_TXCTL				0x06	/* Transition Control */
+#define CS42XX8_DACMUTE				0x07	/* DAC Mute Control */
+#define CS42XX8_VOLAOUT1			0x08	/* Volume Control AOUT1 */
+#define CS42XX8_VOLAOUT2			0x09	/* Volume Control AOUT2 */
+#define CS42XX8_VOLAOUT3			0x0A	/* Volume Control AOUT3 */
+#define CS42XX8_VOLAOUT4			0x0B	/* Volume Control AOUT4 */
+#define CS42XX8_VOLAOUT5			0x0C	/* Volume Control AOUT5 */
+#define CS42XX8_VOLAOUT6			0x0D	/* Volume Control AOUT6 */
+#define CS42XX8_VOLAOUT7			0x0E	/* Volume Control AOUT7 */
+#define CS42XX8_VOLAOUT8			0x0F	/* Volume Control AOUT8 */
+#define CS42XX8_DACINV				0x10	/* DAC Channel Invert */
+#define CS42XX8_VOLAIN1				0x11	/* Volume Control AIN1 */
+#define CS42XX8_VOLAIN2				0x12	/* Volume Control AIN2 */
+#define CS42XX8_VOLAIN3				0x13	/* Volume Control AIN3 */
+#define CS42XX8_VOLAIN4				0x14	/* Volume Control AIN4 */
+#define CS42XX8_VOLAIN5				0x15	/* Volume Control AIN5 */
+#define CS42XX8_VOLAIN6				0x16	/* Volume Control AIN6 */
+#define CS42XX8_ADCINV				0x17	/* ADC Channel Invert */
+#define CS42XX8_STATUSCTL			0x18	/* Status Control */
+#define CS42XX8_STATUS				0x19	/* Status */
+#define CS42XX8_STATUSM				0x1A	/* Status Mask */
+#define CS42XX8_MUTEC				0x1B	/* MUTEC Pin Control */
+
+#define CS42XX8_FIRSTREG			CS42XX8_CHIPID
+#define CS42XX8_LASTREG				CS42XX8_MUTEC
+#define CS42XX8_NUMREGS				(CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1)
+#define CS42XX8_I2C_INCR			0x80
+
+/* Chip I.D. and Revision Register (Address 01h) */
+#define CS42XX8_CHIPID_CHIP_ID_MASK		0xF0
+#define CS42XX8_CHIPID_REV_ID_MASK		0x0F
+
+/* Power Control (Address 02h) */
+#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT		7
+#define CS42XX8_PWRCTL_PDN_ADC3_MASK		(1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC3			(1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT		6
+#define CS42XX8_PWRCTL_PDN_ADC2_MASK		(1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC2			(1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT		5
+#define CS42XX8_PWRCTL_PDN_ADC1_MASK		(1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_ADC1			(1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT		4
+#define CS42XX8_PWRCTL_PDN_DAC4_MASK		(1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC4			(1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT		3
+#define CS42XX8_PWRCTL_PDN_DAC3_MASK		(1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC3			(1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT		2
+#define CS42XX8_PWRCTL_PDN_DAC2_MASK		(1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC2			(1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT		1
+#define CS42XX8_PWRCTL_PDN_DAC1_MASK		(1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_DAC1			(1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT)
+#define CS42XX8_PWRCTL_PDN_SHIFT		0
+#define CS42XX8_PWRCTL_PDN_MASK			(1 << CS42XX8_PWRCTL_PDN_SHIFT)
+#define CS42XX8_PWRCTL_PDN			(1 << CS42XX8_PWRCTL_PDN_SHIFT)
+
+/* Functional Mode (Address 03h) */
+#define CS42XX8_FUNCMOD_DAC_FM_SHIFT		6
+#define CS42XX8_FUNCMOD_DAC_FM_WIDTH		2
+#define CS42XX8_FUNCMOD_DAC_FM_MASK		(((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_DAC_FM(v)		((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_ADC_FM_SHIFT		4
+#define CS42XX8_FUNCMOD_ADC_FM_WIDTH		2
+#define CS42XX8_FUNCMOD_ADC_FM_MASK		(((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_ADC_FM(v)		((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT)
+#define CS42XX8_FUNCMOD_xC_FM_MASK(x)		((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK)
+#define CS42XX8_FUNCMOD_xC_FM(x, v)		((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v))
+#define CS42XX8_FUNCMOD_MFREQ_SHIFT		1
+#define CS42XX8_FUNCMOD_MFREQ_WIDTH		3
+#define CS42XX8_FUNCMOD_MFREQ_MASK		(((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT)
+#define CS42XX8_FUNCMOD_MFREQ_256(s)		((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_384(s)		((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_512(s)		((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_768(s)		((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+#define CS42XX8_FUNCMOD_MFREQ_1024(s)		((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1))
+
+#define CS42XX8_FM_SINGLE			0
+#define CS42XX8_FM_DOUBLE			1
+#define CS42XX8_FM_QUAD				2
+#define CS42XX8_FM_AUTO				3
+
+/* Interface Formats (Address 04h) */
+#define CS42XX8_INTF_FREEZE_SHIFT		7
+#define CS42XX8_INTF_FREEZE_MASK		(1 << CS42XX8_INTF_FREEZE_SHIFT)
+#define CS42XX8_INTF_FREEZE			(1 << CS42XX8_INTF_FREEZE_SHIFT)
+#define CS42XX8_INTF_AUX_DIF_SHIFT		6
+#define CS42XX8_INTF_AUX_DIF_MASK		(1 << CS42XX8_INTF_AUX_DIF_SHIFT)
+#define CS42XX8_INTF_AUX_DIF			(1 << CS42XX8_INTF_AUX_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_SHIFT		3
+#define CS42XX8_INTF_DAC_DIF_WIDTH		3
+#define CS42XX8_INTF_DAC_DIF_MASK		(((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_LEFTJ		(0 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_I2S		(1 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_RIGHTJ		(2 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16		(3 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_ONELINE_20		(4 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_ONELINE_24		(6 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_TDM		(7 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_SHIFT		0
+#define CS42XX8_INTF_ADC_DIF_WIDTH		3
+#define CS42XX8_INTF_ADC_DIF_MASK		(((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_LEFTJ		(0 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_I2S		(1 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_RIGHTJ		(2 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16		(3 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_ONELINE_20		(4 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_ONELINE_24		(6 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_TDM		(7 << CS42XX8_INTF_ADC_DIF_SHIFT)
+
+/* ADC Control & DAC De-Emphasis (Address 05h) */
+#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT	7
+#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK	(1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT)
+#define CS42XX8_ADCCTL_ADC_HPF_FREEZE		(1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT)
+#define CS42XX8_ADCCTL_DAC_DEM_SHIFT		5
+#define CS42XX8_ADCCTL_DAC_DEM_MASK		(1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT)
+#define CS42XX8_ADCCTL_DAC_DEM			(1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT)
+#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT	4
+#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK		(1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC1_SINGLE		(1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT	3
+#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK		(1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC2_SINGLE		(1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT	2
+#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK		(1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_ADC3_SINGLE		(1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT)
+#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT		1
+#define CS42XX8_ADCCTL_AIN5_MUX_MASK		(1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT)
+#define CS42XX8_ADCCTL_AIN5_MUX			(1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT)
+#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT		0
+#define CS42XX8_ADCCTL_AIN6_MUX_MASK		(1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT)
+#define CS42XX8_ADCCTL_AIN6_MUX			(1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT)
+
+/* Transition Control (Address 06h) */
+#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT		7
+#define CS42XX8_TXCTL_DAC_SNGVOL_MASK		(1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_DAC_SNGVOL		(1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_SHIFT		5
+#define CS42XX8_TXCTL_DAC_SZC_WIDTH		2
+#define CS42XX8_TXCTL_DAC_SZC_MASK		(((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_IC		(0 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_ZC		(1 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_SR		(2 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_DAC_SZC_SRZC		(3 << CS42XX8_TXCTL_DAC_SZC_SHIFT)
+#define CS42XX8_TXCTL_AMUTE_SHIFT		4
+#define CS42XX8_TXCTL_AMUTE_MASK		(1 << CS42XX8_TXCTL_AMUTE_SHIFT)
+#define CS42XX8_TXCTL_AMUTE			(1 << CS42XX8_TXCTL_AMUTE_SHIFT)
+#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT		3
+#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK		(1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT)
+#define CS42XX8_TXCTL_MUTE_ADC_SP		(1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT)
+#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT		2
+#define CS42XX8_TXCTL_ADC_SNGVOL_MASK		(1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_ADC_SNGVOL		(1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_SHIFT		0
+#define CS42XX8_TXCTL_ADC_SZC_MASK		(((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_IC		(0 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_ZC		(1 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_SR		(2 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+#define CS42XX8_TXCTL_ADC_SZC_SRZC		(3 << CS42XX8_TXCTL_ADC_SZC_SHIFT)
+
+/* DAC Channel Mute (Address 07h) */
+#define CS42XX8_DACMUTE_AOUT(n)			(0x1 << n)
+#define CS42XX8_DACMUTE_ALL			0xff
+
+/* Status Control (Address 18h)*/
+#define CS42XX8_STATUSCTL_INI_SHIFT		2
+#define CS42XX8_STATUSCTL_INI_WIDTH		2
+#define CS42XX8_STATUSCTL_INI_MASK		(((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT)
+#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH	(0 << CS42XX8_STATUSCTL_INI_SHIFT)
+#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW	(1 << CS42XX8_STATUSCTL_INI_SHIFT)
+#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN	(2 << CS42XX8_STATUSCTL_INI_SHIFT)
+
+/* Status (Address 19h)*/
+#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT	4
+#define CS42XX8_STATUS_DAC_CLK_ERR_MASK		(1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT)
+#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT	3
+#define CS42XX8_STATUS_ADC_CLK_ERR_MASK		(1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT)
+#define CS42XX8_STATUS_ADC3_OVFL_SHIFT		2
+#define CS42XX8_STATUS_ADC3_OVFL_MASK		(1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT)
+#define CS42XX8_STATUS_ADC2_OVFL_SHIFT		1
+#define CS42XX8_STATUS_ADC2_OVFL_MASK		(1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT)
+#define CS42XX8_STATUS_ADC1_OVFL_SHIFT		0
+#define CS42XX8_STATUS_ADC1_OVFL_MASK		(1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT)
+
+/* Status Mask (Address 1Ah) */
+#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT	4
+#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK	(1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT)
+#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT	3
+#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK	(1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT)
+#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT	2
+#define CS42XX8_STATUS_ADC3_OVFL_M_MASK		(1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT)
+#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT	1
+#define CS42XX8_STATUS_ADC2_OVFL_M_MASK		(1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT)
+#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT	0
+#define CS42XX8_STATUS_ADC1_OVFL_M_MASK		(1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT)
+
+/* MUTEC Pin Control (Address 1Bh) */
+#define CS42XX8_MUTEC_MCPOLARITY_SHIFT		1
+#define CS42XX8_MUTEC_MCPOLARITY_MASK		(1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT)
+#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW	(0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT)
+#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH	(1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT)
+#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT	0
+#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK		(1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT)
+#define CS42XX8_MUTEC_MUTEC_ACTIVE		(1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT)
+#endif /* _CS42XX8_H */
-- 
cgit v1.2.3


From 90e8e50fce3585d6f9902701de08389b027dadc6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Mar 2014 19:29:55 -0700
Subject: ASoC: rsnd: add DeviceTree support

Support for loading the Renesas R-Car sound driver via DeviceTree.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/renesas,rsnd.txt     |  96 ++++++++++++++++
 sound/soc/sh/rcar/adg.c                            |   1 +
 sound/soc/sh/rcar/core.c                           | 122 ++++++++++++++++++++-
 sound/soc/sh/rcar/gen.c                            |  15 +++
 sound/soc/sh/rcar/rsnd.h                           |  11 ++
 sound/soc/sh/rcar/src.c                            |  36 ++++++
 sound/soc/sh/rcar/ssi.c                            |  56 ++++++++++
 7 files changed, 334 insertions(+), 3 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.txt

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
new file mode 100644
index 000000000000..7c6d33f29796
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -0,0 +1,96 @@
+Renesas R-Car sound
+
+Required properties:
+- compatible			: "renesas,rcar_sound-gen1" if generation1
+				  "renesas,rcar_sound-gen2" if generation2
+- reg				: Should contain the register physical address.
+				  required register is
+				   SRU/ADG/SSI      if generation1
+				   SRU/ADG/SSIU/SSI if generation2
+- rcar_sound,ssi		: SSI subnode
+- rcar_sound,scu		: SCU subnode
+- rcar_sound,dai		: DAI subnode
+
+SSI subnode properties:
+- interrupts			: Should contain SSI interrupt for PIO transfer
+- shared-pin			: if shared clock pin
+
+DAI subnode properties:
+- playback			: list of playback modules
+- capture			: list of capture  modules
+
+Example:
+
+rcar_sound: rcar_sound@0xffd90000 {
+	#sound-dai-cells = <1>;
+	compatible = "renesas,rcar_sound-gen2";
+	reg =	<0 0xec500000 0 0x1000>, /* SCU */
+		<0 0xec5a0000 0 0x100>,  /* ADG */
+		<0 0xec540000 0 0x1000>, /* SSIU */
+		<0 0xec541000 0 0x1280>; /* SSI */
+
+	rcar_sound,src {
+		src0: src@0 { };
+		src1: src@1 { };
+		src2: src@2 { };
+		src3: src@3 { };
+		src4: src@4 { };
+		src5: src@5 { };
+		src6: src@6 { };
+		src7: src@7 { };
+		src8: src@8 { };
+		src9: src@9 { };
+	};
+
+	rcar_sound,ssi {
+		ssi0: ssi@0 {
+			interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi1: ssi@1 {
+			interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi2: ssi@2 {
+			interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi3: ssi@3 {
+			interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi4: ssi@4 {
+			interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi5: ssi@5 {
+			interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi6: ssi@6 {
+			interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi7: ssi@7 {
+			interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi8: ssi@8 {
+			interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>;
+		};
+		ssi9: ssi@9 {
+			interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>;
+		};
+	};
+
+	rcar_sound,dai {
+		dai0 {
+			playback = <&ssi5 &src5>;
+			capture  = <&ssi6>;
+		};
+		dai1 {
+			playback = <&ssi3>;
+		};
+		dai2 {
+			capture  = <&ssi4>;
+		};
+		dai3 {
+			playback = <&ssi7>;
+		};
+		dai4 {
+			capture  = <&ssi8>;
+		};
+	};
+};
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 953f1cce982d..69c44269ebdb 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -392,6 +392,7 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
 }
 
 int rsnd_adg_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv)
 {
 	struct rsnd_adg *adg;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 6a1b45df8101..e77f7716f1d7 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -100,6 +100,21 @@
 #define RSND_RATES SNDRV_PCM_RATE_8000_96000
 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
 
+static struct rsnd_of_data rsnd_of_data_gen1 = {
+	.flags = RSND_GEN1,
+};
+
+static struct rsnd_of_data rsnd_of_data_gen2 = {
+	.flags = RSND_GEN2,
+};
+
+static struct of_device_id rsnd_of_match[] = {
+	{ .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 },
+	{ .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 },
+	{},
+};
+MODULE_DEVICE_TABLE(of, rsnd_of_match);
+
 /*
  *	rsnd_platform functions
  */
@@ -620,7 +635,92 @@ static int rsnd_path_init(struct rsnd_priv *priv,
 	return ret;
 }
 
+static void rsnd_of_parse_dai(struct platform_device *pdev,
+			      const struct rsnd_of_data *of_data,
+			      struct rsnd_priv *priv)
+{
+	struct device_node *dai_node,	*dai_np;
+	struct device_node *ssi_node,	*ssi_np;
+	struct device_node *src_node,	*src_np;
+	struct device_node *playback, *capture;
+	struct rsnd_dai_platform_info *dai_info;
+	struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+	struct device *dev = &pdev->dev;
+	int nr, i;
+	int dai_i, ssi_i, src_i;
+
+	if (!of_data)
+		return;
+
+	dai_node = of_get_child_by_name(dev->of_node, "rcar_sound,dai");
+	if (!dai_node)
+		return;
+
+	nr = of_get_child_count(dai_node);
+	if (!nr)
+		return;
+
+	dai_info = devm_kzalloc(dev,
+				sizeof(struct rsnd_dai_platform_info) * nr,
+				GFP_KERNEL);
+	if (!dai_info) {
+		dev_err(dev, "dai info allocation error\n");
+		return;
+	}
+
+	info->dai_info_nr	= nr;
+	info->dai_info		= dai_info;
+
+	ssi_node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi");
+	src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src");
+
+#define mod_parse(name)							\
+if (name##_node) {							\
+	struct rsnd_##name##_platform_info *name##_info;		\
+									\
+	name##_i = 0;							\
+	for_each_child_of_node(name##_node, name##_np) {		\
+		name##_info = info->name##_info + name##_i;		\
+									\
+		if (name##_np == playback)				\
+			dai_info->playback.name = name##_info;		\
+		if (name##_np == capture)				\
+			dai_info->capture.name = name##_info;		\
+									\
+		name##_i++;						\
+	}								\
+}
+
+	/*
+	 * parse all dai
+	 */
+	dai_i = 0;
+	for_each_child_of_node(dai_node, dai_np) {
+		dai_info = info->dai_info + dai_i;
+
+		for (i = 0;; i++) {
+
+			playback = of_parse_phandle(dai_np, "playback", i);
+			capture  = of_parse_phandle(dai_np, "capture", i);
+
+			if (!playback && !capture)
+				break;
+
+			mod_parse(ssi);
+			mod_parse(src);
+
+			if (playback)
+				of_node_put(playback);
+			if (capture)
+				of_node_put(capture);
+		}
+
+		dai_i++;
+	}
+}
+
 static int rsnd_dai_probe(struct platform_device *pdev,
+			  const struct rsnd_of_data *of_data,
 			  struct rsnd_priv *priv)
 {
 	struct snd_soc_dai_driver *drv;
@@ -628,13 +728,16 @@ static int rsnd_dai_probe(struct platform_device *pdev,
 	struct rsnd_dai *rdai;
 	struct rsnd_mod *pmod, *cmod;
 	struct device *dev = rsnd_priv_to_dev(priv);
-	int dai_nr = info->dai_info_nr;
+	int dai_nr;
 	int i;
 
+	rsnd_of_parse_dai(pdev, of_data, priv);
+
 	/*
 	 * dai_nr should be set via dai_info_nr,
 	 * but allow it to keeping compatible
 	 */
+	dai_nr = info->dai_info_nr;
 	if (!dai_nr) {
 		/* get max dai nr */
 		for (dai_nr = 0; dai_nr < 32; dai_nr++) {
@@ -802,7 +905,10 @@ static int rsnd_probe(struct platform_device *pdev)
 	struct rsnd_priv *priv;
 	struct device *dev = &pdev->dev;
 	struct rsnd_dai *rdai;
+	const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev);
+	const struct rsnd_of_data *of_data;
 	int (*probe_func[])(struct platform_device *pdev,
+			    const struct rsnd_of_data *of_data,
 			    struct rsnd_priv *priv) = {
 		rsnd_gen_probe,
 		rsnd_ssi_probe,
@@ -812,7 +918,16 @@ static int rsnd_probe(struct platform_device *pdev)
 	};
 	int ret, i;
 
-	info = pdev->dev.platform_data;
+	info = NULL;
+	of_data = NULL;
+	if (of_id) {
+		info = devm_kzalloc(&pdev->dev,
+				    sizeof(struct rcar_snd_info), GFP_KERNEL);
+		of_data = of_id->data;
+	} else {
+		info = pdev->dev.platform_data;
+	}
+
 	if (!info) {
 		dev_err(dev, "driver needs R-Car sound information\n");
 		return -ENODEV;
@@ -835,7 +950,7 @@ static int rsnd_probe(struct platform_device *pdev)
 	 *	init each module
 	 */
 	for (i = 0; i < ARRAY_SIZE(probe_func); i++) {
-		ret = probe_func[i](pdev, priv);
+		ret = probe_func[i](pdev, of_data, priv);
 		if (ret)
 			return ret;
 	}
@@ -903,6 +1018,7 @@ static int rsnd_remove(struct platform_device *pdev)
 static struct platform_driver rsnd_driver = {
 	.driver	= {
 		.name	= "rcar_sound",
+		.of_match_table = rsnd_of_match,
 	},
 	.probe		= rsnd_probe,
 	.remove		= rsnd_remove,
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 9094970dbdfb..50a1ef3eb1c6 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -359,13 +359,28 @@ static int rsnd_gen1_probe(struct platform_device *pdev,
 /*
  *		Gen
  */
+static void rsnd_of_parse_gen(struct platform_device *pdev,
+			      const struct rsnd_of_data *of_data,
+			      struct rsnd_priv *priv)
+{
+	struct rcar_snd_info *info = priv->info;
+
+	if (!of_data)
+		return;
+
+	info->flags = of_data->flags;
+}
+
 int rsnd_gen_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_gen *gen;
 	int ret;
 
+	rsnd_of_parse_gen(pdev, of_data, priv);
+
 	gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
 	if (!gen) {
 		dev_err(dev, "GEN allocate failed\n");
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index c46e0afa54ae..619d198c7d2e 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -17,6 +17,8 @@
 #include <linux/io.h>
 #include <linux/list.h>
 #include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
 #include <linux/sh_dma.h>
 #include <linux/workqueue.h>
 #include <sound/rcar_snd.h>
@@ -113,6 +115,7 @@ enum rsnd_reg {
 #define RSND_REG_SRCOUT_TIMSEL4		RSND_REG_SHARE18
 #define RSND_REG_AUDIO_CLK_SEL2		RSND_REG_SHARE19
 
+struct rsnd_of_data;
 struct rsnd_priv;
 struct rsnd_mod;
 struct rsnd_dai;
@@ -260,6 +263,7 @@ int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional);
  *	R-Car Gen1/Gen2
  */
 int rsnd_gen_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv);
 void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
 			       struct rsnd_mod *mod,
@@ -273,6 +277,7 @@ void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
 int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod);
 int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate);
 int rsnd_adg_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv);
 int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
 				  struct rsnd_mod *mod,
@@ -290,6 +295,10 @@ int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
 /*
  *	R-Car sound priv
  */
+struct rsnd_of_data {
+	u32 flags;
+};
+
 struct rsnd_priv {
 
 	struct device *dev;
@@ -348,6 +357,7 @@ struct rsnd_priv {
  *	R-Car SRC
  */
 int rsnd_src_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id);
 unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv,
@@ -366,6 +376,7 @@ int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod,
  *	R-Car SSI
  */
 int rsnd_ssi_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
 struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index ea6a214985d0..eee75ebf961c 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -628,7 +628,41 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
 	return &((struct rsnd_src *)(priv->src) + id)->mod;
 }
 
+static void rsnd_of_parse_src(struct platform_device *pdev,
+			      const struct rsnd_of_data *of_data,
+			      struct rsnd_priv *priv)
+{
+	struct device_node *src_node;
+	struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+	struct rsnd_src_platform_info *src_info;
+	struct device *dev = &pdev->dev;
+	int nr;
+
+	if (!of_data)
+		return;
+
+	src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src");
+	if (!src_node)
+		return;
+
+	nr = of_get_child_count(src_node);
+	if (!nr)
+		return;
+
+	src_info = devm_kzalloc(dev,
+				sizeof(struct rsnd_src_platform_info) * nr,
+				GFP_KERNEL);
+	if (!src_info) {
+		dev_err(dev, "src info allocation error\n");
+		return;
+	}
+
+	info->src_info		= src_info;
+	info->src_info_nr	= nr;
+}
+
 int rsnd_src_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv)
 {
 	struct rcar_snd_info *info = rsnd_priv_to_info(priv);
@@ -639,6 +673,8 @@ int rsnd_src_probe(struct platform_device *pdev,
 	char name[RSND_SRC_NAME_SIZE];
 	int i, nr;
 
+	rsnd_of_parse_src(pdev, of_data, priv);
+
 	/*
 	 * init SRC
 	 */
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 633b23d209b9..4b7e20603dd7 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -588,7 +588,61 @@ static void rsnd_ssi_parent_clk_setup(struct rsnd_priv *priv, struct rsnd_ssi *s
 	}
 }
 
+
+static void rsnd_of_parse_ssi(struct platform_device *pdev,
+			      const struct rsnd_of_data *of_data,
+			      struct rsnd_priv *priv)
+{
+	struct device_node *node;
+	struct device_node *np;
+	struct rsnd_ssi_platform_info *ssi_info;
+	struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+	struct device *dev = &pdev->dev;
+	int nr, i;
+
+	if (!of_data)
+		return;
+
+	node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi");
+	if (!node)
+		return;
+
+	nr = of_get_child_count(node);
+	if (!nr)
+		return;
+
+	ssi_info = devm_kzalloc(dev,
+				sizeof(struct rsnd_ssi_platform_info) * nr,
+				GFP_KERNEL);
+	if (!ssi_info) {
+		dev_err(dev, "ssi info allocation error\n");
+		return;
+	}
+
+	info->ssi_info		= ssi_info;
+	info->ssi_info_nr	= nr;
+
+	i = -1;
+	for_each_child_of_node(node, np) {
+		i++;
+
+		ssi_info = info->ssi_info + i;
+
+		/*
+		 * pin settings
+		 */
+		if (of_get_property(np, "shared-pin", NULL))
+			ssi_info->flags |= RSND_SSI_CLK_PIN_SHARE;
+
+		/*
+		 * irq
+		 */
+		ssi_info->pio_irq = irq_of_parse_and_map(np, 0);
+	}
+}
+
 int rsnd_ssi_probe(struct platform_device *pdev,
+		   const struct rsnd_of_data *of_data,
 		   struct rsnd_priv *priv)
 {
 	struct rcar_snd_info *info = rsnd_priv_to_info(priv);
@@ -600,6 +654,8 @@ int rsnd_ssi_probe(struct platform_device *pdev,
 	char name[RSND_SSI_NAME_SIZE];
 	int i, nr;
 
+	rsnd_of_parse_ssi(pdev, of_data, priv);
+
 	/*
 	 *	init SSI
 	 */
-- 
cgit v1.2.3


From 8bab0dd58037623b723b768ee2eb1f7dd1ad0416 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Sun, 23 Mar 2014 20:29:15 -0700
Subject: ASoC: rcar: subnode tidyup for renesas,rsnd.txt

rcar_sound,ssi/src/dai subnode documentation
become more cleaner

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 15 ++++++++++++---
 1 file changed, 12 insertions(+), 3 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 7c6d33f29796..a44e9179faf5 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -7,14 +7,23 @@ Required properties:
 				  required register is
 				   SRU/ADG/SSI      if generation1
 				   SRU/ADG/SSIU/SSI if generation2
-- rcar_sound,ssi		: SSI subnode
-- rcar_sound,scu		: SCU subnode
-- rcar_sound,dai		: DAI subnode
+- rcar_sound,ssi		: Should contain SSI feature.
+				  The number of SSI subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,src		: Should contain SRC feature.
+				  The number of SRC subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,dai		: DAI contents.
+				  The number of DAI subnode should be same as HW.
+				  see below for detail.
 
 SSI subnode properties:
 - interrupts			: Should contain SSI interrupt for PIO transfer
 - shared-pin			: if shared clock pin
 
+SRC subnode properties:
+no properties at this point
+
 DAI subnode properties:
 - playback			: list of playback modules
 - capture			: list of capture  modules
-- 
cgit v1.2.3


From 015f630de86c8a79df45c475c34087d3e96b882a Mon Sep 17 00:00:00 2001
From: Jean-Francois Moine <moinejf@free.fr>
Date: Thu, 20 Mar 2014 11:04:16 +0100
Subject: ASoC: simple-card: Add DT documentation for multi-DAI links

Many couples of CPU/CODEC DAI links may be described in the DT
thanks to 'simple-audio-card,dai-link' containers.

Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/simple-card.txt      | 42 +++++++++++++++++++++-
 1 file changed, 41 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index 881914b139ca..131aa2ad7f1a 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -23,6 +23,11 @@ Optional properties:
 
 Required subnodes:
 
+- simple-audio-card,dai-link		: container for the CPU and CODEC sub-nodes
+					  This container may be omitted when the
+					  card has only one DAI link.
+					  See the examples.
+
 - simple-audio-card,cpu			: CPU   sub-node
 - simple-audio-card,codec		: CODEC sub-node
 
@@ -49,7 +54,7 @@ Note:
    CPU and CODEC sides as we need to keep the settings identical for both ends
    of the link.
 
-Example:
+Example 1 - single DAI link:
 
 sound {
 	compatible = "simple-audio-card";
@@ -94,3 +99,38 @@ sh_fsi2: sh_fsi2@ec230000 {
 	interrupt-parent = <&gic>;
 	interrupts = <0 146 0x4>;
 };
+
+Example 2 - many DAI links:
+
+sound {
+	compatible = "simple-audio-card";
+	simple-audio-card,name = "Cubox Audio";
+	simple-audio-card,format = "i2s";
+
+	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
+		simple-audio-card,cpu {
+			sound-dai = <&audio1 0>;
+		};
+		simple-audio-card,codec {
+			sound-dai = <&tda998x 0>;
+		};
+	};
+
+	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
+		simple-audio-card,cpu {
+			sound-dai = <&audio1 1>;
+		};
+		simple-audio-card,codec {
+			sound-dai = <&tda998x 1>;
+		};
+	};
+
+	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
+		simple-audio-card,cpu {
+			sound-dai = <&audio1 1>;
+		};
+		simple-audio-card,codec {
+			sound-dai = <&spdif_codec>;
+		};
+	};
+};
-- 
cgit v1.2.3