Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood', 'asoc/topic/max98090' and 'asoc/topic/mc13783' into asoc-next

Documentation/devicetree/bindings/sound/max98090.txt

@@ -4,7 +4,7 @@ This device supports I2C only.
 
 Required properties:
 
-- compatible : "maxim,max98090".
+- compatible : "maxim,max98090" or "maxim,max98091".
 
 - reg : The I2C address of the device.
 

arch/x86/include/asm/platform_sst_audio.h

@@ -0,0 +1,78 @@
+/*
+ * platform_sst_audio.h:  sst audio platform data header file
+ *
+ * Copyright (C) 2012-14 Intel Corporation
+ * Author: Jeeja KP <jeeja.kp@intel.com>
+ * 	Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
+ *	Vinod Koul ,vinod.koul@intel.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; version 2
+ * of the License.
+ */
+#ifndef _PLATFORM_SST_AUDIO_H_
+#define _PLATFORM_SST_AUDIO_H_
+
+#include <linux/sfi.h>
+
+enum sst_audio_task_id_mrfld {
+	SST_TASK_ID_NONE = 0,
+	SST_TASK_ID_SBA = 1,
+	SST_TASK_ID_MEDIA = 3,
+	SST_TASK_ID_MAX = SST_TASK_ID_MEDIA,
+};
+
+/* Device IDs for Merrifield are Pipe IDs,
+ * ref: DSP spec v0.75 */
+enum sst_audio_device_id_mrfld {
+	/* Output pipeline IDs */
+	PIPE_ID_OUT_START = 0x0,
+	PIPE_CODEC_OUT0 = 0x2,
+	PIPE_CODEC_OUT1 = 0x3,
+	PIPE_SPROT_LOOP_OUT = 0x4,
+	PIPE_MEDIA_LOOP1_OUT = 0x5,
+	PIPE_MEDIA_LOOP2_OUT = 0x6,
+	PIPE_VOIP_OUT = 0xC,
+	PIPE_PCM0_OUT = 0xD,
+	PIPE_PCM1_OUT = 0xE,
+	PIPE_PCM2_OUT = 0xF,
+	PIPE_MEDIA0_OUT = 0x12,
+	PIPE_MEDIA1_OUT = 0x13,
+/* Input Pipeline IDs */
+	PIPE_ID_IN_START = 0x80,
+	PIPE_CODEC_IN0 = 0x82,
+	PIPE_CODEC_IN1 = 0x83,
+	PIPE_SPROT_LOOP_IN = 0x84,
+	PIPE_MEDIA_LOOP1_IN = 0x85,
+	PIPE_MEDIA_LOOP2_IN = 0x86,
+	PIPE_VOIP_IN = 0x8C,
+	PIPE_PCM0_IN = 0x8D,
+	PIPE_PCM1_IN = 0x8E,
+	PIPE_MEDIA0_IN = 0x8F,
+	PIPE_MEDIA1_IN = 0x90,
+	PIPE_MEDIA2_IN = 0x91,
+	PIPE_RSVD = 0xFF,
+};
+
+/* The stream map for each platform consists of an array of the below
+ * stream map structure.
+ */
+struct sst_dev_stream_map {
+	u8 dev_num;		/* device id */
+	u8 subdev_num;		/* substream */
+	u8 direction;
+	u8 device_id;		/* fw id */
+	u8 task_id;		/* fw task */
+	u8 status;
+};
+
+struct sst_platform_data {
+	/* Intel software platform id*/
+	struct sst_dev_stream_map *pdev_strm_map;
+	unsigned int strm_map_size;
+};
+
+int add_sst_platform_device(void);
+#endif
+

include/sound/rt286.h

@@ -0,0 +1,19 @@
+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * 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.
+ */
+
+#ifndef __LINUX_SND_RT286_H
+#define __LINUX_SND_RT286_H
+
+struct rt286_platform_data {
+	bool cbj_en; /*combo jack enable*/
+	bool gpio2_en; /*GPIO2 enable*/
+};
+
+#endif

sound/soc/codecs/Kconfig

@@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM3008
 	select SND_SOC_PCM512x_I2C if I2C
 	select SND_SOC_PCM512x_SPI if SPI_MASTER
+	select SND_SOC_RT286 if I2C
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_RT5640 if I2C
 	select SND_SOC_RT5645 if I2C
@@ -455,6 +456,9 @@ config SND_SOC_RL6231
 	default m if SND_SOC_RT5645=m
 	default m if SND_SOC_RT5651=m
 
+config SND_SOC_RT286
+	tristate
+
 config SND_SOC_RT5631
 	tristate
 

sound/soc/codecs/Makefile

@@ -69,6 +69,7 @@ snd-soc-pcm512x-objs := pcm512x.o
 snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
 snd-soc-pcm512x-spi-objs := pcm512x-spi.o
 snd-soc-rl6231-objs := rl6231.o
+snd-soc-rt286-objs := rt286.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-rt5640-objs := rt5640.o
 snd-soc-rt5645-objs := rt5645.o
@@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_PCM512x)	+= snd-soc-pcm512x.o
 obj-$(CONFIG_SND_SOC_PCM512x_I2C)	+= snd-soc-pcm512x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
 obj-$(CONFIG_SND_SOC_RL6231)	+= snd-soc-rl6231.o
+obj-$(CONFIG_SND_SOC_RT286)	+= snd-soc-rt286.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_RT5640)	+= snd-soc-rt5640.o
 obj-$(CONFIG_SND_SOC_RT5645)	+= snd-soc-rt5645.o

sound/soc/codecs/max98090.c

@@ -26,10 +26,6 @@
 #include <sound/max98090.h>
 #include "max98090.h"
 
-#define DEBUG
-#define EXTMIC_METHOD
-#define EXTMIC_METHOD_TEST
-
 /* Allows for sparsely populated register maps */
 static struct reg_default max98090_reg[] = {
 	{ 0x00, 0x00 }, /* 00 Software Reset */
@@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w,
 	else
 		val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT;
 
-
 	if (val >= 1) {
 		if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) {
 			max98090->pa1en = val - 1; /* Update for volatile */
@@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux =
 	SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
 
 static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
-
 	SND_SOC_DAPM_INPUT("MIC1"),
 	SND_SOC_DAPM_INPUT("MIC2"),
 	SND_SOC_DAPM_INPUT("DMICL"),
@@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
-
 	SND_SOC_DAPM_INPUT("DMIC3"),
 	SND_SOC_DAPM_INPUT("DMIC4"),
 
@@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
-
 	{"MIC1 Input", NULL, "MIC1"},
 	{"MIC2 Input", NULL, "MIC2"},
 
@@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
 	{"SPKR", NULL, "SPK Right Out"},
 	{"RCVL", NULL, "RCV Left Out"},
 	{"RCVR", NULL, "RCV Right Out"},
-
 };
 
 static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
-
 	/* DMIC inputs */
 	{"DMIC3", NULL, "DMIC3_ENA"},
 	{"DMIC4", NULL, "DMIC4_ENA"},
 	{"DMIC3", NULL, "AHPF"},
 	{"DMIC4", NULL, "AHPF"},
-
 };
 
 static int max98090_add_widgets(struct snd_soc_codec *codec)
@@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec)
 
 		snd_soc_dapm_add_routes(dapm, max98091_dapm_routes,
 			ARRAY_SIZE(max98091_dapm_routes));
-
 	}
 
 	return 0;
@@ -2212,22 +2200,11 @@ static struct snd_soc_dai_driver max98090_dai[] = {
 }
 };
 
-static void max98090_handle_pdata(struct snd_soc_codec *codec)
-{
-	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
-	struct max98090_pdata *pdata = max98090->pdata;
-
-	if (!pdata) {
-		dev_err(codec->dev, "No platform data\n");
-		return;
-	}
-
-}
-
 static int max98090_probe(struct snd_soc_codec *codec)
 {
 	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
 	struct max98090_cdata *cdata;
+	enum max98090_type devtype;
 	int ret = 0;
 
 	dev_dbg(codec->dev, "max98090_probe\n");
@@ -2263,16 +2240,21 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	}
 
 	if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) {
-		max98090->devtype = MAX98090;
+		devtype = MAX98090;
 		dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret);
 	} else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) {
-		max98090->devtype = MAX98091;
+		devtype = MAX98091;
 		dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret);
 	} else {
-		max98090->devtype = MAX98090;
+		devtype = MAX98090;
 		dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret);
 	}
 
+	if (max98090->devtype != devtype) {
+		dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n");
+		max98090->devtype = devtype;
+	}
+
 	max98090->jack_state = M98090_JACK_STATE_NO_HEADSET;
 
 	INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work);
@@ -2317,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
 		M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
 
-	max98090_handle_pdata(codec);
-
 	max98090_add_widgets(codec);
 
 err_access:
@@ -2428,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev)
 }
 #endif
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int max98090_resume(struct device *dev)
 {
 	struct max98090_priv *max98090 = dev_get_drvdata(dev);
@@ -2460,12 +2440,14 @@ static const struct dev_pm_ops max98090_pm = {
 
 static const struct i2c_device_id max98090_i2c_id[] = {
 	{ "max98090", MAX98090 },
+	{ "max98091", MAX98091 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, max98090_i2c_id);
 
 static const struct of_device_id max98090_of_match[] = {
 	{ .compatible = "maxim,max98090", },
+	{ .compatible = "maxim,max98091", },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, max98090_of_match);

sound/soc/codecs/mc13783.c

@@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 
 		ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port);
 		if (ret)
-			return ret;
+			goto out;
 
 		ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port);
 		if (ret)
-			return ret;
+			goto out;
 	}
 
 	dev_set_drvdata(&pdev->dev, priv);
@@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
 			mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
 
+out:
+	of_node_put(np);
 	return ret;
 }
 

sound/soc/codecs/rt286.c

@@ -0,0 +1,1224 @@
+/*
+ * rt286.c  --  RT286 ALSA SoC audio codec driver
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <linux/acpi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <linux/workqueue.h>
+#include <sound/rt286.h>
+#include <sound/hda_verbs.h>
+
+#include "rt286.h"
+
+#define RT286_VENDOR_ID 0x10ec0286
+
+struct rt286_priv {
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+	struct rt286_platform_data pdata;
+	struct i2c_client *i2c;
+	struct snd_soc_jack *jack;
+	struct delayed_work jack_detect_work;
+	int sys_clk;
+	struct reg_default *index_cache;
+};
+
+static struct reg_default rt286_index_def[] = {
+	{ 0x01, 0xaaaa },
+	{ 0x02, 0x8aaa },
+	{ 0x03, 0x0002 },
+	{ 0x04, 0xaf01 },
+	{ 0x08, 0x000d },
+	{ 0x09, 0xd810 },
+	{ 0x0a, 0x0060 },
+	{ 0x0b, 0x0000 },
+	{ 0x0d, 0x2800 },
+	{ 0x0f, 0x0000 },
+	{ 0x19, 0x0a17 },
+	{ 0x20, 0x0020 },
+	{ 0x33, 0x0208 },
+	{ 0x49, 0x0004 },
+	{ 0x4f, 0x50e9 },
+	{ 0x50, 0x2c00 },
+	{ 0x63, 0x2902 },
+	{ 0x67, 0x1111 },
+	{ 0x68, 0x1016 },
+	{ 0x69, 0x273f },
+};
+#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def)
+
+static const struct reg_default rt286_reg[] = {
+	{ 0x00170500, 0x00000400 },
+	{ 0x00220000, 0x00000031 },
+	{ 0x00239000, 0x0000007f },
+	{ 0x0023a000, 0x0000007f },
+	{ 0x00270500, 0x00000400 },
+	{ 0x00370500, 0x00000400 },
+	{ 0x00870500, 0x00000400 },
+	{ 0x00920000, 0x00000031 },
+	{ 0x00935000, 0x000000c3 },
+	{ 0x00936000, 0x000000c3 },
+	{ 0x00970500, 0x00000400 },
+	{ 0x00b37000, 0x00000097 },
+	{ 0x00b37200, 0x00000097 },
+	{ 0x00b37300, 0x00000097 },
+	{ 0x00c37000, 0x00000000 },
+	{ 0x00c37100, 0x00000080 },
+	{ 0x01270500, 0x00000400 },
+	{ 0x01370500, 0x00000400 },
+	{ 0x01371f00, 0x411111f0 },
+	{ 0x01439000, 0x00000080 },
+	{ 0x0143a000, 0x00000080 },
+	{ 0x01470700, 0x00000000 },
+	{ 0x01470500, 0x00000400 },
+	{ 0x01470c00, 0x00000000 },
+	{ 0x01470100, 0x00000000 },
+	{ 0x01837000, 0x00000000 },
+	{ 0x01870500, 0x00000400 },
+	{ 0x02050000, 0x00000000 },
+	{ 0x02139000, 0x00000080 },
+	{ 0x0213a000, 0x00000080 },
+	{ 0x02170100, 0x00000000 },
+	{ 0x02170500, 0x00000400 },
+	{ 0x02170700, 0x00000000 },
+	{ 0x02270100, 0x00000000 },
+	{ 0x02370100, 0x00000000 },
+	{ 0x02040000, 0x00004002 },
+	{ 0x01870700, 0x00000020 },
+	{ 0x00830000, 0x000000c3 },
+	{ 0x00930000, 0x000000c3 },
+	{ 0x01270700, 0x00000000 },
+};
+
+static bool rt286_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case 0 ... 0xff:
+	case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+	case RT286_GET_HP_SENSE:
+	case RT286_GET_MIC1_SENSE:
+	case RT286_PROC_COEF:
+		return true;
+	default:
+		return false;
+	}
+
+
+}
+
+static bool rt286_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case 0 ... 0xff:
+	case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+	case RT286_GET_HP_SENSE:
+	case RT286_GET_MIC1_SENSE:
+	case RT286_SET_AUDIO_POWER:
+	case RT286_SET_HPO_POWER:
+	case RT286_SET_SPK_POWER:
+	case RT286_SET_DMIC1_POWER:
+	case RT286_SPK_MUX:
+	case RT286_HPO_MUX:
+	case RT286_ADC0_MUX:
+	case RT286_ADC1_MUX:
+	case RT286_SET_MIC1:
+	case RT286_SET_PIN_HPO:
+	case RT286_SET_PIN_SPK:
+	case RT286_SET_PIN_DMIC1:
+	case RT286_SPK_EAPD:
+	case RT286_SET_AMP_GAIN_HPO:
+	case RT286_SET_DMIC2_DEFAULT:
+	case RT286_DACL_GAIN:
+	case RT286_DACR_GAIN:
+	case RT286_ADCL_GAIN:
+	case RT286_ADCR_GAIN:
+	case RT286_MIC_GAIN:
+	case RT286_SPOL_GAIN:
+	case RT286_SPOR_GAIN:
+	case RT286_HPOL_GAIN:
+	case RT286_HPOR_GAIN:
+	case RT286_F_DAC_SWITCH:
+	case RT286_F_RECMIX_SWITCH:
+	case RT286_REC_MIC_SWITCH:
+	case RT286_REC_I2S_SWITCH:
+	case RT286_REC_LINE_SWITCH:
+	case RT286_REC_BEEP_SWITCH:
+	case RT286_DAC_FORMAT:
+	case RT286_ADC_FORMAT:
+	case RT286_COEF_INDEX:
+	case RT286_PROC_COEF:
+	case RT286_SET_AMP_GAIN_ADC_IN1:
+	case RT286_SET_AMP_GAIN_ADC_IN2:
+	case RT286_SET_POWER(RT286_DAC_OUT1):
+	case RT286_SET_POWER(RT286_DAC_OUT2):
+	case RT286_SET_POWER(RT286_ADC_IN1):
+	case RT286_SET_POWER(RT286_ADC_IN2):
+	case RT286_SET_POWER(RT286_DMIC2):
+	case RT286_SET_POWER(RT286_MIC1):
+		return true;
+	default:
+		return false;
+	}
+}
+
+static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
+{
+	struct i2c_client *client = context;
+	struct rt286_priv *rt286 = i2c_get_clientdata(client);
+	u8 data[4];
+	int ret, i;
+
+	/*handle index registers*/
+	if (reg <= 0xff) {
+		rt286_hw_write(client, RT286_COEF_INDEX, reg);
+		reg = RT286_PROC_COEF;
+		for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+			if (reg == rt286->index_cache[i].reg) {
+				rt286->index_cache[i].def = value;
+				break;
+			}
+
+		}
+	}
+
+	data[0] = (reg >> 24) & 0xff;
+	data[1] = (reg >> 16) & 0xff;
+	/*
+	 * 4 bit VID: reg should be 0
+	 * 12 bit VID: value should be 0
+	 * So we use an OR operator to handle it rather than use if condition.
+	 */
+	data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff);
+	data[3] = value & 0xff;
+
+	ret = i2c_master_send(client, data, 4);
+
+	if (ret == 4)
+		return 0;
+	else
+		pr_err("ret=%d\n", ret);
+	if (ret < 0)
+		return ret;
+	else
+		return -EIO;
+}
+
+static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value)
+{
+	struct i2c_client *client = context;
+	struct i2c_msg xfer[2];
+	int ret;
+	__be32 be_reg;
+	unsigned int index, vid, buf = 0x0;
+
+	/*handle index registers*/
+	if (reg <= 0xff) {
+		rt286_hw_write(client, RT286_COEF_INDEX, reg);
+		reg = RT286_PROC_COEF;
+	}
+
+	reg = reg | 0x80000;
+	vid = (reg >> 8) & 0xfff;
+
+	if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) {
+		index = (reg >> 8) & 0xf;
+		reg = (reg & ~0xf0f) | index;
+	}
+	be_reg = cpu_to_be32(reg);
+
+	/* Write register */
+	xfer[0].addr = client->addr;
+	xfer[0].flags = 0;
+	xfer[0].len = 4;
+	xfer[0].buf = (u8 *)&be_reg;
+
+	/* Read data */
+	xfer[1].addr = client->addr;
+	xfer[1].flags = I2C_M_RD;
+	xfer[1].len = 4;
+	xfer[1].buf = (u8 *)&buf;
+
+	ret = i2c_transfer(client->adapter, xfer, 2);
+	if (ret < 0)
+		return ret;
+	else if (ret != 2)
+		return -EIO;
+
+	*value = be32_to_cpu(buf);
+
+	return 0;
+}
+
+static void rt286_index_sync(struct snd_soc_codec *codec)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+		snd_soc_write(codec, rt286->index_cache[i].reg,
+				  rt286->index_cache[i].def);
+	}
+}
+
+static int rt286_support_power_controls[] = {
+	RT286_DAC_OUT1,
+	RT286_DAC_OUT2,
+	RT286_ADC_IN1,
+	RT286_ADC_IN2,
+	RT286_MIC1,
+	RT286_DMIC1,
+	RT286_DMIC2,
+	RT286_SPK_OUT,
+	RT286_HP_OUT,
+};
+#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls)
+
+static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val, buf;
+	int i;
+
+	*hp = false;
+	*mic = false;
+
+	if (rt286->pdata.cbj_en) {
+		buf = snd_soc_read(codec, RT286_GET_HP_SENSE);
+		*hp = buf & 0x80000000;
+		if (*hp) {
+			/* power on HV,VERF */
+			snd_soc_update_bits(codec,
+				RT286_POWER_CTRL1, 0x1001, 0x0);
+			/* power LDO1 */
+			snd_soc_update_bits(codec,
+				RT286_POWER_CTRL2, 0x4, 0x4);
+			snd_soc_write(codec, RT286_SET_MIC1, 0x24);
+			val = snd_soc_read(codec, RT286_CBJ_CTRL2);
+
+			msleep(200);
+			i = 40;
+			while (((val & 0x0800) == 0) && (i > 0)) {
+				val =  snd_soc_read(codec,
+					RT286_CBJ_CTRL2);
+				i--;
+				msleep(20);
+			}
+
+			if (0x0400 == (val & 0x0700)) {
+				*mic = false;
+
+				snd_soc_write(codec,
+					RT286_SET_MIC1, 0x20);
+				/* power off HV,VERF */
+				snd_soc_update_bits(codec,
+					RT286_POWER_CTRL1, 0x1001, 0x1001);
+				snd_soc_update_bits(codec,
+					RT286_A_BIAS_CTRL3, 0xc000, 0x0000);
+				snd_soc_update_bits(codec,
+					RT286_CBJ_CTRL1, 0x0030, 0x0000);
+				snd_soc_update_bits(codec,
+					RT286_A_BIAS_CTRL2, 0xc000, 0x0000);
+			} else if ((0x0200 == (val & 0x0700)) ||
+				(0x0100 == (val & 0x0700))) {
+				*mic = true;
+				snd_soc_update_bits(codec,
+					RT286_A_BIAS_CTRL3, 0xc000, 0x8000);
+				snd_soc_update_bits(codec,
+					RT286_CBJ_CTRL1, 0x0030, 0x0020);
+				snd_soc_update_bits(codec,
+					RT286_A_BIAS_CTRL2, 0xc000, 0x8000);
+			} else {
+				*mic = false;
+			}
+
+			snd_soc_update_bits(codec,
+						RT286_MISC_CTRL1,
+						0x0060, 0x0000);
+		} else {
+			snd_soc_update_bits(codec,
+						RT286_MISC_CTRL1,
+						0x0060, 0x0020);
+			snd_soc_update_bits(codec,
+						RT286_A_BIAS_CTRL3,
+						0xc000, 0x8000);
+			snd_soc_update_bits(codec,
+						RT286_CBJ_CTRL1,
+						0x0030, 0x0020);
+			snd_soc_update_bits(codec,
+						RT286_A_BIAS_CTRL2,
+						0xc000, 0x8000);
+
+			*mic = false;
+		}
+	} else {
+		buf = snd_soc_read(codec, RT286_GET_HP_SENSE);
+		*hp = buf & 0x80000000;
+		buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE);
+		*mic = buf & 0x80000000;
+	}
+
+	return 0;
+}
+
+static void rt286_jack_detect_work(struct work_struct *work)
+{
+	struct rt286_priv *rt286 =
+		container_of(work, struct rt286_priv, jack_detect_work.work);
+	int status = 0;
+	bool hp = false;
+	bool mic = false;
+
+	rt286_jack_detect(rt286->codec, &hp, &mic);
+
+	if (hp == true)
+		status |= SND_JACK_HEADPHONE;
+
+	if (mic == true)
+		status |= SND_JACK_MICROPHONE;
+
+	snd_soc_jack_report(rt286->jack, status,
+		SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+}
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+	rt286->jack = jack;
+
+	/* Send an initial empty report */
+	snd_soc_jack_report(rt286->jack, 0,
+		SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt286_mic_detect);
+
+static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0);
+
+static const struct snd_kcontrol_new rt286_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN,
+			    RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+	SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN,
+			    RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+	SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN,
+			    0, 0x3, 0, mic_vol_tlv),
+	SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN,
+			    RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1),
+};
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt286_front_mix[] = {
+	SOC_DAPM_SINGLE("DAC Switch",  RT286_F_DAC_SWITCH,
+			RT286_MUTE_SFT, 1, 1),
+	SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH,
+			RT286_MUTE_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt286_rec_mix[] = {
+	SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH,
+			RT286_MUTE_SFT, 1, 1),
+	SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH,
+			RT286_MUTE_SFT, 1, 1),
+	SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH,
+			RT286_MUTE_SFT, 1, 1),
+	SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH,
+			RT286_MUTE_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new spo_enable_control =
+	SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK,
+			RT286_SET_PIN_SFT, 1, 0);
+
+static const struct snd_kcontrol_new hpol_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN,
+			RT286_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hpor_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN,
+			RT286_MUTE_SFT, 1, 1);
+
+/* ADC0 source */
+static const char * const rt286_adc_src[] = {
+	"Mic", "RECMIX", "Dmic"
+};
+
+static const int rt286_adc_values[] = {
+	0, 4, 5,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(
+	rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT,
+	RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values);
+
+static const struct snd_kcontrol_new rt286_adc0_mux =
+	SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(
+	rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT,
+	RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values);
+
+static const struct snd_kcontrol_new rt286_adc1_mux =
+	SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum);
+
+static const char * const rt286_dac_src[] = {
+	"Front", "Surround"
+};
+/* HP-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX,
+				0, rt286_dac_src);
+
+static const struct snd_kcontrol_new rt286_hpo_mux =
+SOC_DAPM_ENUM("HPO source", rt286_hpo_enum);
+
+/* SPK-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX,
+				0, rt286_dac_src);
+
+static const struct snd_kcontrol_new rt286_spo_mux =
+SOC_DAPM_ENUM("SPO source", rt286_spo_enum);
+
+static int rt286_spk_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_write(codec,
+			RT286_SPK_EAPD, RT286_SET_EAPD_HIGH);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_write(codec,
+			RT286_SPK_EAPD, RT286_SET_EAPD_LOW);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt286_adc_event(struct snd_soc_dapm_widget *w,
+			     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int nid;
+
+	nid = (w->reg >> 20) & 0xff;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_update_bits(codec,
+			VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
+			0x7080, 0x7000);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(codec,
+			VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
+			0x7080, 0x7080);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
+	/* Input Lines */
+	SND_SOC_DAPM_INPUT("DMIC1 Pin"),
+	SND_SOC_DAPM_INPUT("DMIC2 Pin"),
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_INPUT("LINE1"),
+	SND_SOC_DAPM_INPUT("Beep"),
+
+	/* DMIC */
+	SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1,
+		NULL, 0, rt286_set_dmic1_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1,
+		NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM,
+		0, 0, NULL, 0),
+
+	/* REC Mixer */
+	SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0,
+		rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0),
+
+	/* ADC Mux */
+	SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1,
+		&rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
+		SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1,
+		&rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
+		SND_SOC_DAPM_POST_PMU),
+
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+	/* Output Side */
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0),
+
+	/* Output Mux */
+	SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux),
+	SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux),
+
+	SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO,
+		RT286_SET_PIN_SFT, 0, NULL, 0),
+
+	/* Output Mixer */
+	SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1,
+			rt286_front_mix, ARRAY_SIZE(rt286_front_mix)),
+	SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1,
+			NULL, 0),
+
+	/* Output Pga */
+	SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0,
+		&spo_enable_control, rt286_spk_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0,
+		&hpol_enable_control),
+	SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0,
+		&hpor_enable_control),
+
+	/* Output Lines */
+	SND_SOC_DAPM_OUTPUT("SPOL"),
+	SND_SOC_DAPM_OUTPUT("SPOR"),
+	SND_SOC_DAPM_OUTPUT("HPO Pin"),
+	SND_SOC_DAPM_OUTPUT("SPDIF"),
+};
+
+static const struct snd_soc_dapm_route rt286_dapm_routes[] = {
+	{"DMIC1", NULL, "DMIC1 Pin"},
+	{"DMIC2", NULL, "DMIC2 Pin"},
+	{"DMIC1", NULL, "DMIC Receiver"},
+	{"DMIC2", NULL, "DMIC Receiver"},
+
+	{"RECMIX", "Beep Switch", "Beep"},
+	{"RECMIX", "Line1 Switch", "LINE1"},
+	{"RECMIX", "Mic1 Switch", "MIC1"},
+
+	{"ADC 0 Mux", "Dmic", "DMIC1"},
+	{"ADC 0 Mux", "RECMIX", "RECMIX"},
+	{"ADC 0 Mux", "Mic", "MIC1"},
+	{"ADC 1 Mux", "Dmic", "DMIC2"},
+	{"ADC 1 Mux", "RECMIX", "RECMIX"},
+	{"ADC 1 Mux", "Mic", "MIC1"},
+
+	{"ADC 0", NULL, "ADC 0 Mux"},
+	{"ADC 1", NULL, "ADC 1 Mux"},
+
+	{"AIF1TX", NULL, "ADC 0"},
+	{"AIF2TX", NULL, "ADC 1"},
+
+	{"DAC 0", NULL, "AIF1RX"},
+	{"DAC 1", NULL, "AIF2RX"},
+
+	{"Front", "DAC Switch", "DAC 0"},
+	{"Front", "RECMIX Switch", "RECMIX"},
+
+	{"Surround", NULL, "DAC 1"},
+
+	{"SPK Mux", "Front", "Front"},
+	{"SPK Mux", "Surround", "Surround"},
+
+	{"HPO Mux", "Front", "Front"},
+	{"HPO Mux", "Surround", "Surround"},
+
+	{"SPO", "Switch", "SPK Mux"},
+	{"HPO L", "Switch", "HPO Mux"},
+	{"HPO R", "Switch", "HPO Mux"},
+	{"HPO L", NULL, "HP Power"},
+	{"HPO R", NULL, "HP Power"},
+
+	{"SPOL", NULL, "SPO"},
+	{"SPOR", NULL, "SPO"},
+	{"HPO Pin", NULL, "HPO L"},
+	{"HPO Pin", NULL, "HPO R"},
+};
+
+static int rt286_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val = 0;
+	int d_len_code;
+
+	switch (params_rate(params)) {
+	/* bit 14 0:48K 1:44.1K */
+	case 44100:
+		val |= 0x4000;
+		break;
+	case 48000:
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported sample rate %d\n",
+					params_rate(params));
+		return -EINVAL;
+	}
+	switch (rt286->sys_clk) {
+	case 12288000:
+	case 24576000:
+		if (params_rate(params) != 48000) {
+			dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+					params_rate(params), rt286->sys_clk);
+			return -EINVAL;
+		}
+		break;
+	case 11289600:
+	case 22579200:
+		if (params_rate(params) != 44100) {
+			dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+					params_rate(params), rt286->sys_clk);
+			return -EINVAL;
+		}
+		break;
+	}
+
+	if (params_channels(params) <= 16) {
+		/* bit 3:0 Number of Channel */
+		val |= (params_channels(params) - 1);
+	} else {
+		dev_err(codec->dev, "Unsupported channels %d\n",
+					params_channels(params));
+		return -EINVAL;
+	}
+
+	d_len_code = 0;
+	switch (params_width(params)) {
+	/* bit 6:4 Bits per Sample */
+	case 16:
+		d_len_code = 0;
+		val |= (0x1 << 4);
+		break;
+	case 32:
+		d_len_code = 2;
+		val |= (0x4 << 4);
+		break;
+	case 20:
+		d_len_code = 1;
+		val |= (0x2 << 4);
+		break;
+	case 24:
+		d_len_code = 2;
+		val |= (0x3 << 4);
+		break;
+	case 8:
+		d_len_code = 3;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec,
+		RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
+	dev_dbg(codec->dev, "format val = 0x%x\n", val);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+	else
+		snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+
+	return 0;
+}
+
+static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x800, 0x800);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x800, 0x0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x300, 0x0);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x300, 0x1 << 8);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x300, 0x2 << 8);
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x300, 0x3 << 8);
+		break;
+	default:
+		return -EINVAL;
+	}
+	/* bit 15 Stream Type 0:PCM 1:Non-PCM */
+	snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0);
+	snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0);
+
+	return 0;
+}
+
+static int rt286_set_dai_sysclk(struct snd_soc_dai *dai,
+				int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq);
+
+	if (RT286_SCLK_S_MCLK == clk_id) {
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL2, 0x0100, 0x0);
+		snd_soc_update_bits(codec,
+			RT286_PLL_CTRL1, 0x20, 0x20);
+	} else {
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL2, 0x0100, 0x0100);
+		snd_soc_update_bits(codec,
+			RT286_PLL_CTRL, 0x4, 0x4);
+		snd_soc_update_bits(codec,
+			RT286_PLL_CTRL1, 0x20, 0x0);
+	}
+
+	switch (freq) {
+	case 19200000:
+		if (RT286_SCLK_S_MCLK == clk_id) {
+			dev_err(codec->dev, "Should not use MCLK\n");
+			return -EINVAL;
+		}
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL2, 0x40, 0x40);
+		break;
+	case 24000000:
+		if (RT286_SCLK_S_MCLK == clk_id) {
+			dev_err(codec->dev, "Should not use MCLK\n");
+			return -EINVAL;
+		}
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL2, 0x40, 0x0);
+		break;
+	case 12288000:
+	case 11289600:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL2, 0x8, 0x0);
+		snd_soc_update_bits(codec,
+			RT286_CLK_DIV, 0xfc1e, 0x0004);
+		break;
+	case 24576000:
+	case 22579200:
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL2, 0x8, 0x8);
+		snd_soc_update_bits(codec,
+			RT286_CLK_DIV, 0xfc1e, 0x5406);
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported system clock\n");
+		return -EINVAL;
+	}
+
+	rt286->sys_clk = freq;
+
+	return 0;
+}
+
+static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio);
+	if (50 == ratio)
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x1000, 0x1000);
+	else
+		snd_soc_update_bits(codec,
+			RT286_I2S_CTRL1, 0x1000, 0x0);
+
+
+	return 0;
+}
+
+static int rt286_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+			snd_soc_write(codec,
+				RT286_SET_AUDIO_POWER, AC_PWRST_D0);
+			snd_soc_update_bits(codec,
+				RT286_DC_GAIN, 0x200, 0x200);
+		}
+		break;
+
+	case SND_SOC_BIAS_ON:
+		mdelay(10);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_write(codec,
+			RT286_SET_AUDIO_POWER, AC_PWRST_D3);
+		snd_soc_update_bits(codec,
+			RT286_DC_GAIN, 0x200, 0x0);
+		break;
+
+	default:
+		break;
+	}
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static irqreturn_t rt286_irq(int irq, void *data)
+{
+	struct rt286_priv *rt286 = data;
+	bool hp = false;
+	bool mic = false;
+	int status = 0;
+
+	rt286_jack_detect(rt286->codec, &hp, &mic);
+
+	/* Clear IRQ */
+	snd_soc_update_bits(rt286->codec,
+					RT286_IRQ_CTRL, 0x1, 0x1);
+
+	if (hp == true)
+		status |= SND_JACK_HEADPHONE;
+
+	if (mic == true)
+		status |= SND_JACK_MICROPHONE;
+
+	snd_soc_jack_report(rt286->jack, status,
+		SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+	pm_wakeup_event(&rt286->i2c->dev, 300);
+
+	return IRQ_HANDLED;
+}
+
+static int rt286_probe(struct snd_soc_codec *codec)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+	codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+	rt286->codec = codec;
+
+	return 0;
+}
+
+static int rt286_remove(struct snd_soc_codec *codec)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+	cancel_delayed_work_sync(&rt286->jack_detect_work);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int rt286_suspend(struct snd_soc_codec *codec)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+	regcache_cache_only(rt286->regmap, true);
+	regcache_mark_dirty(rt286->regmap);
+
+	return 0;
+}
+
+static int rt286_resume(struct snd_soc_codec *codec)
+{
+	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+	regcache_cache_only(rt286->regmap, false);
+	rt286_index_sync(codec);
+	regcache_sync(rt286->regmap);
+
+	return 0;
+}
+#else
+#define rt286_suspend NULL
+#define rt286_resume NULL
+#endif
+
+#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt286_aif_dai_ops = {
+	.hw_params = rt286_hw_params,
+	.set_fmt = rt286_set_dai_fmt,
+	.set_sysclk = rt286_set_dai_sysclk,
+	.set_bclk_ratio = rt286_set_bclk_ratio,
+};
+
+static struct snd_soc_dai_driver rt286_dai[] = {
+	{
+		.name = "rt286-aif1",
+		.id = RT286_AIF1,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT286_STEREO_RATES,
+			.formats = RT286_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT286_STEREO_RATES,
+			.formats = RT286_FORMATS,
+		},
+		.ops = &rt286_aif_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "rt286-aif2",
+		.id = RT286_AIF2,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT286_STEREO_RATES,
+			.formats = RT286_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF2 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT286_STEREO_RATES,
+			.formats = RT286_FORMATS,
+		},
+		.ops = &rt286_aif_dai_ops,
+		.symmetric_rates = 1,
+	},
+
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_rt286 = {
+	.probe = rt286_probe,
+	.remove = rt286_remove,
+	.suspend = rt286_suspend,
+	.resume = rt286_resume,
+	.set_bias_level = rt286_set_bias_level,
+	.idle_bias_off = true,
+	.controls = rt286_snd_controls,
+	.num_controls = ARRAY_SIZE(rt286_snd_controls),
+	.dapm_widgets = rt286_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets),
+	.dapm_routes = rt286_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes),
+};
+
+static const struct regmap_config rt286_regmap = {
+	.reg_bits = 32,
+	.val_bits = 32,
+	.max_register = 0x02370100,
+	.volatile_reg = rt286_volatile_register,
+	.readable_reg = rt286_readable_register,
+	.reg_write = rt286_hw_write,
+	.reg_read = rt286_hw_read,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = rt286_reg,
+	.num_reg_defaults = ARRAY_SIZE(rt286_reg),
+};
+
+static const struct i2c_device_id rt286_i2c_id[] = {
+	{"rt286", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, rt286_i2c_id);
+
+static const struct acpi_device_id rt286_acpi_match[] = {
+	{ "INT343A", 0 },
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, rt286_acpi_match);
+
+static int rt286_i2c_probe(struct i2c_client *i2c,
+			   const struct i2c_device_id *id)
+{
+	struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev);
+	struct rt286_priv *rt286;
+	int i, ret;
+
+	rt286 = devm_kzalloc(&i2c->dev,	sizeof(*rt286),
+				GFP_KERNEL);
+	if (NULL == rt286)
+		return -ENOMEM;
+
+	rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap);
+	if (IS_ERR(rt286->regmap)) {
+		ret = PTR_ERR(rt286->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	regmap_read(rt286->regmap,
+		RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret);
+	if (ret != RT286_VENDOR_ID) {
+		dev_err(&i2c->dev,
+			"Device with ID register %x is not rt286\n", ret);
+		return -ENODEV;
+	}
+
+	rt286->index_cache = rt286_index_def;
+	rt286->i2c = i2c;
+	i2c_set_clientdata(i2c, rt286);
+
+	if (pdata)
+		rt286->pdata = *pdata;
+
+	regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
+
+	for (i = 0; i < RT286_POWER_REG_LEN; i++)
+		regmap_write(rt286->regmap,
+			RT286_SET_POWER(rt286_support_power_controls[i]),
+			AC_PWRST_D1);
+
+	if (!rt286->pdata.cbj_en) {
+		regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000);
+		regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816);
+		regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000);
+		regmap_update_bits(rt286->regmap,
+					RT286_CBJ_CTRL1, 0xf000, 0xb000);
+	} else {
+		regmap_update_bits(rt286->regmap,
+					RT286_CBJ_CTRL1, 0xf000, 0x5000);
+	}
+
+	mdelay(10);
+
+	if (!rt286->pdata.gpio2_en)
+		regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000);
+	else
+		regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0);
+
+	mdelay(10);
+
+	/*Power down LDO2*/
+	regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0);
+
+	/*Set depop parameter*/
+	regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a);
+	regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
+	regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
+
+	if (rt286->i2c->irq) {
+		regmap_update_bits(rt286->regmap,
+					RT286_IRQ_CTRL, 0x2, 0x2);
+
+		INIT_DELAYED_WORK(&rt286->jack_detect_work,
+					rt286_jack_detect_work);
+		schedule_delayed_work(&rt286->jack_detect_work,
+					msecs_to_jiffies(1250));
+
+		ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq,
+			IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286);
+		if (ret != 0) {
+			dev_err(&i2c->dev,
+				"Failed to reguest IRQ: %d\n", ret);
+			return ret;
+		}
+	}
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286,
+				     rt286_dai, ARRAY_SIZE(rt286_dai));
+
+	return ret;
+}
+
+static int rt286_i2c_remove(struct i2c_client *i2c)
+{
+	struct rt286_priv *rt286 = i2c_get_clientdata(i2c);
+
+	if (i2c->irq)
+		free_irq(i2c->irq, rt286);
+	snd_soc_unregister_codec(&i2c->dev);
+
+	return 0;
+}
+
+
+static struct i2c_driver rt286_i2c_driver = {
+	.driver = {
+		   .name = "rt286",
+		   .owner = THIS_MODULE,
+		   .acpi_match_table = ACPI_PTR(rt286_acpi_match),
+		   },
+	.probe = rt286_i2c_probe,
+	.remove = rt286_i2c_remove,
+	.id_table = rt286_i2c_id,
+};
+
+module_i2c_driver(rt286_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT286 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/rt286.h

@@ -0,0 +1,198 @@
+/*
+ * rt286.h  --  RT286 ALSA SoC audio driver
+ *
+ * Copyright 2011 Realtek Microelectronics
+ * Author: Johnny Hsu <johnnyhsu@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __RT286_H__
+#define __RT286_H__
+
+#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
+
+#define RT286_AUDIO_FUNCTION_GROUP			0x01
+#define RT286_DAC_OUT1					0x02
+#define RT286_DAC_OUT2					0x03
+#define RT286_ADC_IN1					0x09
+#define RT286_ADC_IN2					0x08
+#define RT286_MIXER_IN					0x0b
+#define RT286_MIXER_OUT1				0x0c
+#define RT286_MIXER_OUT2				0x0d
+#define RT286_DMIC1					0x12
+#define RT286_DMIC2					0x13
+#define RT286_SPK_OUT					0x14
+#define RT286_MIC1					0x18
+#define RT286_LINE1					0x1a
+#define RT286_BEEP					0x1d
+#define RT286_SPDIF					0x1e
+#define RT286_VENDOR_REGISTERS				0x20
+#define RT286_HP_OUT					0x21
+#define RT286_MIXER_IN1					0x22
+#define RT286_MIXER_IN2					0x23
+
+#define RT286_SET_PIN_SFT				6
+#define RT286_SET_PIN_ENABLE				0x40
+#define RT286_SET_PIN_DISABLE				0
+#define RT286_SET_EAPD_HIGH				0x2
+#define RT286_SET_EAPD_LOW				0
+
+#define RT286_MUTE_SFT					7
+
+/* Verb commands */
+#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM)
+#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0)
+#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP)
+#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT)
+#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT)
+#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1)
+#define RT286_SPK_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0)
+#define RT286_HPO_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0)
+#define RT286_ADC0_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0)
+#define RT286_ADC1_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0)
+#define RT286_SET_MIC1\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0)
+#define RT286_SET_PIN_HPO\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0)
+#define RT286_SET_PIN_SPK\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0)
+#define RT286_SET_PIN_DMIC1\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0)
+#define RT286_SPK_EAPD\
+	VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0)
+#define RT286_SET_AMP_GAIN_HPO\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN1\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN2\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0)
+#define RT286_GET_HP_SENSE\
+	VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0)
+#define RT286_GET_MIC1_SENSE\
+	VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0)
+#define RT286_SET_DMIC2_DEFAULT\
+	VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0)
+#define RT286_DACL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000)
+#define RT286_DACR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000)
+#define RT286_ADCL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000)
+#define RT286_ADCR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000)
+#define RT286_MIC_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000)
+#define RT286_SPOL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000)
+#define RT286_SPOR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000)
+#define RT286_HPOL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000)
+#define RT286_HPOR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000)
+#define RT286_F_DAC_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000)
+#define RT286_F_RECMIX_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100)
+#define RT286_REC_MIC_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000)
+#define RT286_REC_I2S_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100)
+#define RT286_REC_LINE_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200)
+#define RT286_REC_BEEP_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300)
+#define RT286_DAC_FORMAT\
+	VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0)
+#define RT286_ADC_FORMAT\
+	VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0)
+#define RT286_COEF_INDEX\
+	VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
+#define RT286_PROC_COEF\
+	VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
+
+/* Index registers */
+#define RT286_A_BIAS_CTRL1	0x01
+#define RT286_A_BIAS_CTRL2	0x02
+#define RT286_POWER_CTRL1	0x03
+#define RT286_A_BIAS_CTRL3	0x04
+#define RT286_POWER_CTRL2	0x08
+#define RT286_I2S_CTRL1		0x09
+#define RT286_I2S_CTRL2		0x0a
+#define RT286_CLK_DIV		0x0b
+#define RT286_DC_GAIN		0x0d
+#define RT286_POWER_CTRL3	0x0f
+#define RT286_MIC1_DET_CTRL	0x19
+#define RT286_MISC_CTRL1	0x20
+#define RT286_IRQ_CTRL		0x33
+#define RT286_PLL_CTRL1		0x49
+#define RT286_CBJ_CTRL1		0x4f
+#define RT286_CBJ_CTRL2		0x50
+#define RT286_PLL_CTRL		0x63
+#define RT286_DEPOP_CTRL1	0x66
+#define RT286_DEPOP_CTRL2	0x67
+#define RT286_DEPOP_CTRL3	0x68
+#define RT286_DEPOP_CTRL4	0x69
+
+/* SPDIF (0x06) */
+#define RT286_SPDIF_SEL_SFT	0
+#define RT286_SPDIF_SEL_PCM0	0
+#define RT286_SPDIF_SEL_PCM1	1
+#define RT286_SPDIF_SEL_SPOUT	2
+#define RT286_SPDIF_SEL_PP	3
+
+/* RECMIX (0x0b) */
+#define RT286_M_REC_BEEP_SFT	0
+#define RT286_M_REC_LINE1_SFT	1
+#define RT286_M_REC_MIC1_SFT	2
+#define RT286_M_REC_I2S_SFT	3
+
+/* Front (0x0c) */
+#define RT286_M_FRONT_DAC_SFT	0
+#define RT286_M_FRONT_REC_SFT	1
+
+/* SPK-OUT (0x14) */
+#define RT286_M_SPK_MUX_SFT	14
+#define RT286_SPK_SEL_MASK	0x1
+#define RT286_SPK_SEL_SFT	0
+#define RT286_SPK_SEL_F		0
+#define RT286_SPK_SEL_S		1
+
+/* HP-OUT (0x21) */
+#define RT286_M_HP_MUX_SFT	14
+#define RT286_HP_SEL_MASK	0x1
+#define RT286_HP_SEL_SFT	0
+#define RT286_HP_SEL_F		0
+#define RT286_HP_SEL_S		1
+
+/* ADC (0x22) (0x23) */
+#define RT286_ADC_SEL_MASK	0x7
+#define RT286_ADC_SEL_SFT	0
+#define RT286_ADC_SEL_SURR	0
+#define RT286_ADC_SEL_FRONT	1
+#define RT286_ADC_SEL_DMIC	2
+#define RT286_ADC_SEL_BEEP	4
+#define RT286_ADC_SEL_LINE1	5
+#define RT286_ADC_SEL_I2S	6
+#define RT286_ADC_SEL_MIC1	7
+
+#define RT286_SCLK_S_MCLK	0
+#define RT286_SCLK_S_PLL	1
+
+enum {
+	RT286_AIF1,
+	RT286_AIF2,
+	RT286_AIFS,
+};
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+#endif /* __RT286_H__ */
+

sound/soc/intel/Kconfig

@@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
 	help
 	  This adds audio driver for Intel Baytrail platform based boards
 	  with the MAX98090 audio codec.
+
+config SND_SOC_INTEL_BROADWELL_MACH
+	tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
+	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
+	select SND_SOC_INTEL_HASWELL
+	select SND_COMPRESS_OFFLOAD
+	select SND_SOC_RT286
+	help
+	  This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
+	  Ultrabook platforms.
+	  Say Y if you have such a device
+	  If unsure select "N".

sound/soc/intel/Makefile

@@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o
 snd-soc-sst-haswell-objs := haswell.o
 snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
 snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-broadwell-objs := broadwell.o
 
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o

sound/soc/intel/broadwell.c

@@ -0,0 +1,251 @@
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 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/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt286.h"
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("DMIC1", NULL),
+	SND_SOC_DAPM_MIC("DMIC2", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack deteck */
+	{"Headphones", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+	{"LINE1", NULL, "Line Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC1"},
+	{"DMIC2 Pin", NULL, "DMIC2"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The ADSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+		SND_SOC_CLOCK_IN);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+	.hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+	struct sst_hsw *broadwell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings */
+	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+		SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	if (ret < 0) {
+		dev_err(rtd->dev, "error: failed to set device config\n");
+		return ret;
+	}
+
+	/* always connected - check HP for jack detect */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line Jack");
+	snd_soc_dapm_enable_pin(dapm, "DMIC1");
+	snd_soc_dapm_enable_pin(dapm, "DMIC2");
+
+	return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System PCM",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = broadwell_rtd_init,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload0",
+		.stream_name = "Offload0 Playback",
+		.cpu_dai_name = "Offload0 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload1",
+		.stream_name = "Offload1 Playback",
+		.cpu_dai_name = "Offload1 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Loopback PCM",
+		.stream_name = "Loopback",
+		.cpu_dai_name = "Loopback Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 0,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "Capture PCM",
+		.stream_name = "Capture",
+		.cpu_dai_name = "Capture Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "Codec",
+		.be_id = 0,
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt286-aif1",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broadwell_ssp0_fixup,
+		.ops = &broadwell_rt286_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+	.name = "broadwell-rt286",
+	.owner = THIS_MODULE,
+	.dai_link = broadwell_rt286_dais,
+	.num_links = ARRAY_SIZE(broadwell_rt286_dais),
+	.dapm_widgets = broadwell_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+	.dapm_routes = broadwell_rt286_map,
+	.num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+	.fully_routed = true,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+	broadwell_rt286.dev = &pdev->dev;
+
+	return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&broadwell_rt286);
+	return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+	.probe = broadwell_audio_probe,
+	.remove = broadwell_audio_remove,
+	.driver = {
+		.name = "broadwell-audio",
+		.owner = THIS_MODULE,
+	},
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");

sound/soc/intel/byt-max98090.c

@@ -63,14 +63,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = {
 		.pin	= "Headset Mic",
 		.mask	= SND_JACK_MICROPHONE,
 	},
-	{
-		.pin	= "Ext Spk",
-		.mask	= SND_JACK_LINEOUT,
-	},
-	{
-		.pin	= "Int Mic",
-		.mask	= SND_JACK_LINEIN,
-	},
 };
 
 static struct snd_soc_jack_gpio hs_jack_gpios[] = {

sound/soc/intel/byt-rt5640.c

@@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+	{"Headset Mic", NULL, "MICBIAS1"},
 	{"IN2P", NULL, "Headset Mic"},
 	{"IN2N", NULL, "Headset Mic"},
 	{"DMIC1", NULL, "Internal Mic"},

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

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (C) 2013-14 Intel Corp
+ *  Author: Ramesh Babu <ramesh.babu.koul@intel.com>
+ *  	Omair M Abdullah <omair.m.abdullah@intel.com>
+ *  	Samreen Nilofer <samreen.nilofer@intel.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; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+
+#ifndef __SST_CONTROLS_V2_H__
+#define __SST_CONTROLS_V2_H__
+
+enum {
+	MERR_DPCM_AUDIO = 0,
+	MERR_DPCM_COMPR,
+};
+
+
+#endif

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

@@ -122,6 +122,26 @@ struct sst_byt_tstamp {
 	u32 channel_peak[8];
 } __packed;
 
+struct sst_byt_fw_version {
+	u8 build;
+	u8 minor;
+	u8 major;
+	u8 type;
+} __packed;
+
+struct sst_byt_fw_build_info {
+	u8 date[16];
+	u8 time[16];
+} __packed;
+
+struct sst_byt_fw_init {
+	struct sst_byt_fw_version fw_version;
+	struct sst_byt_fw_build_info build_info;
+	u16 result;
+	u8 module_id;
+	u8 debug_info;
+} __packed;
+
 /* driver internal IPC message structure */
 struct ipc_message {
 	struct list_head list;
@@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_byt *byt;
 	struct sst_fw *byt_sst_fw;
+	struct sst_byt_fw_init init;
 	int err;
 
 	dev_dbg(dev, "initialising Byt DSP IPC\n");
@@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 		goto boot_err;
 	}
 
+	/* show firmware information */
+	sst_dsp_inbox_read(byt->dsp, &init, sizeof(init));
+	dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n",
+		 init.fw_version.major, init.fw_version.minor,
+		 init.fw_version.build, init.fw_version.type);
+	dev_info(byt->dev, "Build type: %x\n", init.fw_version.type);
+	dev_info(byt->dev, "Build date: %s %s\n",
+		 init.build_info.date, init.build_info.time);
+
 	pdata->dsp = byt;
 	byt->fw = byt_sst_fw;
 

sound/soc/intel/sst-dsp.c

@@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64);
 
 void sst_dsp_dump(struct sst_dsp *sst)
 {
-	sst->ops->dump(sst);
+	if (sst->ops->dump)
+		sst->ops->dump(sst);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_dump);
 
 void sst_dsp_reset(struct sst_dsp *sst)
 {
-	sst->ops->reset(sst);
+	if (sst->ops->reset)
+		sst->ops->reset(sst);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_reset);
 
 int sst_dsp_boot(struct sst_dsp *sst)
 {
-	sst->ops->boot(sst);
+	if (sst->ops->boot)
+		sst->ops->boot(sst);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(sst_dsp_boot);

sound/soc/intel/sst-dsp.h

@@ -52,7 +52,11 @@
 #define SST_CLKCTL		0x78
 #define SST_CSR2		0x80
 #define SST_LTRC		0xE0
-#define SST_HDMC		0xE8
+#define SST_HMDC		0xE8
+
+#define SST_SHIM_BEGIN		SST_CSR
+#define SST_SHIM_END		SST_HDMC
+
 #define SST_DBGO		0xF0
 
 #define SST_SHIM_SIZE		0x100
@@ -73,6 +77,8 @@
 #define SST_CSR_S0IOCS		(0x1 << 21)
 #define SST_CSR_S1IOCS		(0x1 << 23)
 #define SST_CSR_LPCS		(0x1 << 31)
+#define SST_CSR_24MHZ_LPCS	(SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS)
+#define SST_CSR_24MHZ_NO_LPCS	(SST_CSR_SBCS0 | SST_CSR_SBCS1)
 #define SST_BYT_CSR_RST		(0x1 << 0)
 #define SST_BYT_CSR_VECTOR_SEL	(0x1 << 1)
 #define SST_BYT_CSR_STALL	(0x1 << 2)
@@ -92,6 +98,14 @@
 #define SST_IMRX_DONE		(0x1 << 0)
 #define SST_BYT_IMRX_REQUEST	(0x1 << 1)
 
+/* IMRD / IMD */
+#define SST_IMRD_DONE		(0x1 << 0)
+#define SST_IMRD_BUSY		(0x1 << 1)
+#define SST_IMRD_SSP0		(0x1 << 16)
+#define SST_IMRD_DMAC0		(0x1 << 21)
+#define SST_IMRD_DMAC1		(0x1 << 22)
+#define SST_IMRD_DMAC		(SST_IMRD_DMAC0 | SST_IMRD_DMAC1)
+
 /*  IPCX / IPCC */
 #define	SST_IPCX_DONE		(0x1 << 30)
 #define	SST_IPCX_BUSY		(0x1 << 31)
@@ -118,9 +132,21 @@
 /* LTRC */
 #define SST_LTRC_VAL(x)		(x << 0)
 
-/* HDMC */
-#define SST_HDMC_HDDA0(x)	(x << 0)
-#define SST_HDMC_HDDA1(x)	(x << 7)
+/* HMDC */
+#define SST_HMDC_HDDA0(x)	(x << 0)
+#define SST_HMDC_HDDA1(x)	(x << 7)
+#define SST_HMDC_HDDA_E0_CH0	1
+#define SST_HMDC_HDDA_E0_CH1	2
+#define SST_HMDC_HDDA_E0_CH2	4
+#define SST_HMDC_HDDA_E0_CH3	8
+#define SST_HMDC_HDDA_E1_CH0	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0)
+#define SST_HMDC_HDDA_E1_CH1	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1)
+#define SST_HMDC_HDDA_E1_CH2	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2)
+#define SST_HMDC_HDDA_E1_CH3	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E0_ALLCH	(SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \
+				 SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E1_ALLCH	(SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \
+				 SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3)
 
 
 /* SST Vendor Defined Registers and bits */
@@ -130,11 +156,16 @@
 #define SST_VDRTCTL3		0xaC
 
 /* VDRTCTL0 */
+#define SST_VDRTCL0_APLLSE_MASK		1
 #define SST_VDRTCL0_DSRAMPGE_SHIFT	16
 #define SST_VDRTCL0_DSRAMPGE_MASK	(0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
 #define SST_VDRTCL0_ISRAMPGE_SHIFT	6
 #define SST_VDRTCL0_ISRAMPGE_MASK	(0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
 
+/* PMCS */
+#define SST_PMCS		0x84
+#define SST_PMCS_PS_MASK	0x3
+
 struct sst_dsp;
 
 /*

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

@@ -28,9 +28,6 @@
 #include <linux/firmware.h>
 #include <linux/pm_runtime.h>
 
-#include <linux/acpi.h>
-#include <acpi/acpi_bus.h>
-
 #include "sst-dsp.h"
 #include "sst-dsp-priv.h"
 #include "sst-haswell-ipc.h"
@@ -272,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst)
 		SST_CSR2_SDFD_SSP1);
 
 	/* enable DMA engine 0,1 all channels to access host memory */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
-		SST_HDMC_HDDA1(0xff)  | SST_HDMC_HDDA0(0xff),
-		SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+	sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC,
+		SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff),
+		SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff));
 
 	/* disable all clock gating */
 	writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
@@ -313,9 +310,7 @@ static const struct sst_adsp_memregion lp_region[] = {
 
 /* wild cat point ADSP mem regions */
 static const struct sst_adsp_memregion wpt_region[] = {
-	{0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
-	{0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
-	{0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+	{0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */
 	{0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
 };
 
@@ -339,21 +334,40 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
 	return 0;
 }
 
+struct sst_sram_shift {
+	u32 dev_id;	/* SST Device IDs  */
+	u32 iram_shift;
+	u32 dram_shift;
+};
+
+static const struct sst_sram_shift sram_shift[] = {
+	{SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
+	{SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
+};
 static u32 hsw_block_get_bit(struct sst_mem_block *block)
 {
-	u32 bit = 0, shift = 0;
+	u32 bit = 0, shift = 0, index;
+	struct sst_dsp *sst = block->dsp;
 
-	switch (block->type) {
-	case SST_MEM_DRAM:
-		shift = 16;
-		break;
-	case SST_MEM_IRAM:
-		shift = 6;
-		break;
-	default:
-		return 0;
+	for (index = 0; index < ARRAY_SIZE(sram_shift); index++) {
+		if (sram_shift[index].dev_id == sst->id)
+			break;
 	}
 
+	if (index < ARRAY_SIZE(sram_shift)) {
+		switch (block->type) {
+		case SST_MEM_DRAM:
+			shift = sram_shift[index].dram_shift;
+			break;
+		case SST_MEM_IRAM:
+			shift = sram_shift[index].iram_shift;
+			break;
+		default:
+			shift = 0;
+		}
+	} else
+		shift = 0;
+
 	bit = 1 << (block->index + shift);
 
 	return bit;
@@ -501,8 +515,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 		}
 	}
 
-	/* set default power gating mask */
-	writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+	/* set default power gating control, enable power gating control for all blocks. that is,
+	can't be accessed, please enable each block before accessing. */
+	writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
 
 	return 0;
 }

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

@@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready {
 	u32 inbox_size;
 	u32 outbox_size;
 	u32 fw_info_size;
-	u8 fw_info[1];
+	u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
 } __attribute__((packed));
 
 struct ipc_message {
@@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work)
 		return;
 	}
 
-	/* if the DSP is busy we will TX messages after IRQ */
+	/* if the DSP is busy, we will TX messages after IRQ.
+	 * also postpone if we are in the middle of procesing completion irq*/
 	ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
-	if (ipcx & SST_IPCX_BUSY) {
+	if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) {
 		spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
 		return;
 	}
@@ -502,6 +503,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg,
 		ipc_shim_dbg(hsw, "message timeout");
 
 		trace_ipc_error("error message timeout for", msg->header);
+		list_del(&msg->list);
 		ret = -ETIMEDOUT;
 	} else {
 
@@ -569,6 +571,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
 {
 	struct sst_hsw_ipc_fw_ready fw_ready;
 	u32 offset;
+	u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
+	char *tmp[5], *pinfo;
+	int i = 0;
 
 	offset = (header & 0x1FFFFFFF) << 3;
 
@@ -589,6 +594,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
 		fw_ready.inbox_offset, fw_ready.inbox_size);
 	dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
 		fw_ready.outbox_offset, fw_ready.outbox_size);
+	if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) {
+		fw_ready.fw_info[fw_ready.fw_info_size] = 0;
+		dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info);
+
+		/* log the FW version info got from the mailbox here. */
+		memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size);
+		pinfo = &fw_info[0];
+		for (i = 0; i < sizeof(tmp) / sizeof(char *); i++)
+			tmp[i] = strsep(&pinfo, " ");
+		dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - "
+			"version: %s.%s, build %s, source commit id: %s\n",
+			tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]);
+	}
 }
 
 static void hsw_notification_work(struct work_struct *work)
@@ -671,7 +689,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
 	switch (stream_msg) {
 	case IPC_STR_STAGE_MESSAGE:
 	case IPC_STR_NOTIFICATION:
+		break;
 	case IPC_STR_RESET:
+		trace_ipc_notification("stream reset", stream->reply.stream_hw_id);
 		break;
 	case IPC_STR_PAUSE:
 		stream->running = false;
@@ -762,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
 	}
 
 	/* update any stream states */
-	hsw_stream_update(hsw, msg);
+	if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE)
+		hsw_stream_update(hsw, msg);
 
 	/* wake up and return the error if we have waiters on this message ? */
 	list_del(&msg->list);
@@ -1628,7 +1649,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
 {
 	u32 header, state_;
-	int ret;
+	int ret, item;
 
 	header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
 	state_ = state;
@@ -1642,6 +1663,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 		return ret;
 	}
 
+	for (item = 0; item < dx->entries_no; item++) {
+		dev_dbg(hsw->dev,
+			"Item[%d] offset[%x] - size[%x] - source[%x]\n",
+			item, dx->mem_info[item].offset,
+			dx->mem_info[item].size,
+			dx->mem_info[item].source);
+	}
 	dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
 		dx->entries_no, state);
 
@@ -1775,8 +1803,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
 
 	/* get the FW version */
 	sst_hsw_fw_get_version(hsw, &version);
-	dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
-		version.type, version.major, version.minor, version.build);
 
 	/* get the globalmixer */
 	ret = sst_hsw_mixer_get_info(hsw);

sound/soc/intel/sst-mfld-dsp.h

@@ -3,7 +3,7 @@
 /*
  *  sst_mfld_dsp.h - Intel SST Driver for audio engine
  *
- *  Copyright (C) 2008-12 Intel Corporation
+ *  Copyright (C) 2008-14 Intel Corporation
  *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
  *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  *
@@ -19,6 +19,142 @@
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */
 
+#define SST_MAX_BIN_BYTES 1024
+
+#define MAX_DBG_RW_BYTES 80
+#define MAX_NUM_SCATTER_BUFFERS 8
+#define MAX_LOOP_BACK_DWORDS 8
+/* IPC base address and mailbox, timestamp offsets */
+#define SST_MAILBOX_SIZE 0x0400
+#define SST_MAILBOX_SEND 0x0000
+#define SST_TIME_STAMP 0x1800
+#define SST_TIME_STAMP_MRFLD 0x800
+#define SST_RESERVED_OFFSET 0x1A00
+#define SST_SCU_LPE_MAILBOX 0x1000
+#define SST_LPE_SCU_MAILBOX 0x1400
+#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16)
+#define PROCESS_MSG 0x80
+
+/* Message ID's for IPC messages */
+/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */
+
+/* I2L Firmware/Codec Download msgs */
+#define IPC_IA_PREP_LIB_DNLD 0x01
+#define IPC_IA_LIB_DNLD_CMPLT 0x02
+#define IPC_IA_GET_FW_VERSION 0x04
+#define IPC_IA_GET_FW_BUILD_INF 0x05
+#define IPC_IA_GET_FW_INFO 0x06
+#define IPC_IA_GET_FW_CTXT 0x07
+#define IPC_IA_SET_FW_CTXT 0x08
+#define IPC_IA_PREPARE_SHUTDOWN 0x31
+/* I2L Codec Config/control msgs */
+#define IPC_PREP_D3 0x10
+#define IPC_IA_SET_CODEC_PARAMS 0x10
+#define IPC_IA_GET_CODEC_PARAMS 0x11
+#define IPC_IA_SET_PPP_PARAMS 0x12
+#define IPC_IA_GET_PPP_PARAMS 0x13
+#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA
+#define IPC_IA_ALG_PARAMS 0x1A
+#define IPC_IA_TUNING_PARAMS 0x1B
+#define IPC_IA_SET_RUNTIME_PARAMS 0x1C
+#define IPC_IA_SET_PARAMS 0x1
+#define IPC_IA_GET_PARAMS 0x2
+
+#define IPC_EFFECTS_CREATE 0xE
+#define IPC_EFFECTS_DESTROY 0xF
+
+/* I2L Stream config/control msgs */
+#define IPC_IA_ALLOC_STREAM_MRFLD 0x2
+#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */
+#define IPC_IA_FREE_STREAM_MRFLD 0x03
+#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */
+#define IPC_IA_SET_STREAM_PARAMS 0x22
+#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12
+#define IPC_IA_GET_STREAM_PARAMS 0x23
+#define IPC_IA_PAUSE_STREAM 0x24
+#define IPC_IA_PAUSE_STREAM_MRFLD 0x4
+#define IPC_IA_RESUME_STREAM 0x25
+#define IPC_IA_RESUME_STREAM_MRFLD 0x5
+#define IPC_IA_DROP_STREAM 0x26
+#define IPC_IA_DROP_STREAM_MRFLD 0x07
+#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */
+#define IPC_IA_DRAIN_STREAM_MRFLD 0x8
+#define IPC_IA_CONTROL_ROUTING 0x29
+#define IPC_IA_VTSV_UPDATE_MODULES 0x20
+#define IPC_IA_VTSV_DETECTED 0x21
+
+#define IPC_IA_START_STREAM_MRFLD 0X06
+#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */
+
+#define IPC_IA_SET_GAIN_MRFLD 0x21
+/* Debug msgs */
+#define IPC_IA_DBG_MEM_READ 0x40
+#define IPC_IA_DBG_MEM_WRITE 0x41
+#define IPC_IA_DBG_LOOP_BACK 0x42
+#define IPC_IA_DBG_LOG_ENABLE 0x45
+#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47
+
+/* L2I Firmware/Codec Download msgs */
+#define IPC_IA_FW_INIT_CMPLT 0x81
+#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01
+#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11
+
+/* L2I Codec Config/control msgs */
+#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */
+
+#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */
+#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */
+#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */
+#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */
+#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */
+#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */
+
+#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */
+/* L2S messages */
+#define IPC_SC_DDR_LINK_UP 0xC0
+#define IPC_SC_DDR_LINK_DOWN 0xC1
+#define IPC_SC_SET_LPECLK_REQ 0xC2
+#define IPC_SC_SSP_BIT_BANG 0xC3
+
+/* L2I Error reporting msgs */
+#define IPC_IA_MEM_ALLOC_FAIL 0xE0
+#define IPC_IA_PROC_ERR 0xE1 /* error in processing a
+					stream can be used by playback and
+					capture modules */
+
+/* L2I Debug msgs */
+#define IPC_IA_PRINT_STRING 0xF0
+
+/* Buffer under-run */
+#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B
+
+/* Mrfld specific defines:
+ * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR)
+ * received from FW, the format is:
+ *  - IPC High: pvt_id is set to zero. Always short message.
+ *  - msg_id is in lower 16-bits of IPC low payload.
+ *  - pipe_id is in higher 16-bits of IPC low payload for period_elapsed.
+ *  - error id is in higher 16-bits of IPC low payload for async errors.
+ */
+#define SST_ASYNC_DRV_ID 0
+
+/* Command Response or Acknowledge message to any IPC message will have
+ * same message ID and stream ID information which is sent.
+ * There is no specific Ack message ID. The data field is used as response
+ * meaning.
+ */
+enum ackData {
+	IPC_ACK_SUCCESS = 0,
+	IPC_ACK_FAILURE,
+};
+
+enum ipc_ia_msg_id {
+	IPC_CMD = 1,		/*!< Task Control message ID */
+	IPC_SET_PARAMS = 2,/*!< Task Set param message ID */
+	IPC_GET_PARAMS = 3,	/*!< Task Get param message ID */
+	IPC_INVALID = 0xFF,	/*!<Task Get param message ID */
+};
+
 enum sst_codec_types {
 	/*  AUDIO/MUSIC	CODEC Type Definitions */
 	SST_CODEC_TYPE_UNKNOWN = 0,
@@ -35,14 +171,157 @@ enum stream_type {
 	SST_STREAM_TYPE_MUSIC = 1,
 };
 
+enum sst_error_codes {
+	/* Error code,response to msgId: Description */
+	/* Common error codes */
+	SST_SUCCESS = 0,        /* Success */
+	SST_ERR_INVALID_STREAM_ID = 1,
+	SST_ERR_INVALID_MSG_ID = 2,
+	SST_ERR_INVALID_STREAM_OP = 3,
+	SST_ERR_INVALID_PARAMS = 4,
+	SST_ERR_INVALID_CODEC = 5,
+	SST_ERR_INVALID_MEDIA_TYPE = 6,
+	SST_ERR_STREAM_ERR = 7,
+
+	SST_ERR_STREAM_IN_USE = 15,
+};
+
+struct ipc_dsp_hdr {
+	u16 mod_index_id:8;		/*!< DSP Command ID specific to tasks */
+	u16 pipe_id:8;	/*!< instance of the module in the pipeline */
+	u16 mod_id;		/*!< Pipe_id */
+	u16 cmd_id;		/*!< Module ID = lpe_algo_types_t */
+	u16 length;		/*!< Length of the payload only */
+} __packed;
+
+union ipc_header_high {
+	struct {
+		u32  msg_id:8;	    /* Message ID - Max 256 Message Types */
+		u32  task_id:4;	    /* Task ID associated with this comand */
+		u32  drv_id:4;    /* Identifier for the driver to track*/
+		u32  rsvd1:8;	    /* Reserved */
+		u32  result:4;	    /* Reserved */
+		u32  res_rqd:1;	    /* Response rqd */
+		u32  large:1;	    /* Large Message if large = 1 */
+		u32  done:1;	    /* bit 30 - Done bit */
+		u32  busy:1;	    /* bit 31 - busy bit*/
+	} part;
+	u32 full;
+} __packed;
+/* IPC header */
+union ipc_header_mrfld {
+	struct {
+		u32 header_low_payload;
+		union ipc_header_high header_high;
+	} p;
+	u64 full;
+} __packed;
+/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/
+
+/* IPC Header */
+union ipc_header {
+	struct {
+		u32  msg_id:8; /* Message ID - Max 256 Message Types */
+		u32  str_id:5;
+		u32  large:1;	/* Large Message if large = 1 */
+		u32  reserved:2;	/* Reserved for future use */
+		u32  data:14;	/* Ack/Info for msg, size of msg in Mailbox */
+		u32  done:1; /* bit 30 */
+		u32  busy:1; /* bit 31 */
+	} part;
+	u32 full;
+} __packed;
+
+/* Firmware build info */
+struct sst_fw_build_info {
+	unsigned char  date[16]; /* Firmware build date */
+	unsigned char  time[16]; /* Firmware build time */
+} __packed;
+
+/* Firmware Version info */
+struct snd_sst_fw_version {
+	u8 build;	/* build number*/
+	u8 minor;	/* minor number*/
+	u8 major;	/* major number*/
+	u8 type;	/* build type */
+};
+
+struct ipc_header_fw_init {
+	struct snd_sst_fw_version fw_version;/* Firmware version details */
+	struct sst_fw_build_info build_info;
+	u16 result;	/* Fw init result */
+	u8 module_id; /* Module ID in case of error */
+	u8 debug_info; /* Debug info from Module ID in case of fail */
+} __packed;
+
+struct snd_sst_tstamp {
+	u64 ring_buffer_counter;	/* PB/CP: Bytes copied from/to DDR. */
+	u64 hardware_counter;	    /* PB/CP: Bytes DMAed to/from SSP. */
+	u64 frames_decoded;
+	u64 bytes_decoded;
+	u64 bytes_copied;
+	u32 sampling_frequency;
+	u32 channel_peak[8];
+} __packed;
+
+/* Stream type params struture for Alloc stream */
+struct snd_sst_str_type {
+	u8 codec_type;		/* Codec type */
+	u8 str_type;		/* 1 = voice 2 = music */
+	u8 operation;		/* Playback or Capture */
+	u8 protected_str;	/* 0=Non DRM, 1=DRM */
+	u8 time_slots;
+	u8 reserved;		/* Reserved */
+	u16 result;		/* Result used for acknowledgment */
+} __packed;
+
+/* Library info structure */
+struct module_info {
+	u32 lib_version;
+	u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/
+	u32 media_type;
+	u8  lib_name[12];
+	u32 lib_caps;
+	unsigned char  b_date[16]; /* Lib build date */
+	unsigned char  b_time[16]; /* Lib build time */
+} __packed;
+
+/* Library slot info */
+struct lib_slot_info {
+	u8  slot_num; /* 1 or 2 */
+	u8  reserved1;
+	u16 reserved2;
+	u32 iram_size; /* slot size in IRAM */
+	u32 dram_size; /* slot size in DRAM */
+	u32 iram_offset; /* starting offset of slot in IRAM */
+	u32 dram_offset; /* starting offset of slot in DRAM */
+} __packed;
+
+struct snd_ppp_mixer_params {
+	__u32			type; /*Type of the parameter */
+	__u32			size;
+	__u32			input_stream_bitmap; /*Input stream Bit Map*/
+} __packed;
+
+struct snd_sst_lib_download {
+	struct module_info lib_info; /* library info type, capabilities etc */
+	struct lib_slot_info slot_info; /* slot info to be downloaded */
+	u32 mod_entry_pt;
+};
+
+struct snd_sst_lib_download_info {
+	struct snd_sst_lib_download dload_lib;
+	u16 result;	/* Result used for acknowledgment */
+	u8 pvt_id; /* Private ID */
+	u8 reserved;  /* for alignment */
+};
 struct snd_pcm_params {
 	u8 num_chan;	/* 1=Mono, 2=Stereo */
 	u8 pcm_wd_sz;	/* 16/24 - bit*/
-	u32 reserved;	/* Bitrate in bits per second */
-	u32 sfreq;	/* Sampling rate in Hz */
-	u8 use_offload_path;
+	u8 use_offload_path;	/* 0-PCM using period elpased & ALSA interfaces
+				   1-PCM stream via compressed interface  */
 	u8 reserved2;
-	u16 reserved3;
+	u32 sfreq;    /* Sampling rate in Hz */
 	u8 channel_map[8];
 } __packed;
 
@@ -76,6 +355,7 @@ struct snd_aac_params {
 struct snd_wma_params {
 	u8  num_chan;	/* 1=Mono, 2=Stereo */
 	u8  pcm_wd_sz;	/* 16/24 - bit*/
+	u16 reserved1;
 	u32 brate;	/* Use the hard coded value. */
 	u32 sfreq;	/* Sampling freq eg. 8000, 441000, 48000 */
 	u32 channel_mask;  /* Channel Mask */
@@ -101,26 +381,153 @@ struct sst_address_info {
 };
 
 struct snd_sst_alloc_params_ext {
-	struct sst_address_info  ring_buf_info[8];
-	u8 sg_count;
-	u8 reserved;
-	u16 reserved2;
-	u32 frag_size;	/*Number of samples after which period elapsed
+	__u16 sg_count;
+	__u16 reserved;
+	__u32 frag_size;	/*Number of samples after which period elapsed
 				  message is sent valid only if path  = 0*/
-} __packed;
+	struct sst_address_info  ring_buf_info[8];
+};
 
 struct snd_sst_stream_params {
 	union snd_sst_codec_params uc;
 } __packed;
 
 struct snd_sst_params {
+	u32 result;
 	u32 stream_id;
 	u8 codec;
 	u8 ops;
 	u8 stream_type;
 	u8 device_type;
+	u8 task;
 	struct snd_sst_stream_params sparams;
 	struct snd_sst_alloc_params_ext aparams;
 };
 
+struct snd_sst_alloc_mrfld {
+	u16 codec_type;
+	u8 operation;
+	u8 sg_count;
+	struct sst_address_info ring_buf_info[8];
+	u32 frag_size;
+	u32 ts;
+	struct snd_sst_stream_params codec_params;
+} __packed;
+
+/* Alloc stream params structure */
+struct snd_sst_alloc_params {
+	struct snd_sst_str_type str_type;
+	struct snd_sst_stream_params stream_params;
+	struct snd_sst_alloc_params_ext alloc_params;
+} __packed;
+
+/* Alloc stream response message */
+struct snd_sst_alloc_response {
+	struct snd_sst_str_type str_type; /* Stream type for allocation */
+	struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */
+};
+
+/* Drop response */
+struct snd_sst_drop_response {
+	u32 result;
+	u32 bytes;
+};
+
+struct snd_sst_async_msg {
+	u32 msg_id; /* Async msg id */
+	u32 payload[0];
+};
+
+struct snd_sst_async_err_msg {
+	u32 fw_resp; /* Firmware Result */
+	u32 lib_resp; /*Library result */
+} __packed;
+
+struct snd_sst_vol {
+	u32	stream_id;
+	s32	volume;
+	u32	ramp_duration;
+	u32	ramp_type;		/* Ramp type, default=0 */
+};
+
+/* Gain library parameters for mrfld
+ * based on DSP command spec v0.82
+ */
+struct snd_sst_gain_v2 {
+	u16 gain_cell_num;  /* num of gain cells to modify*/
+	u8 cell_nbr_idx; /* instance index*/
+	u8 cell_path_idx; /* pipe-id */
+	u16 module_id; /*module id */
+	u16 left_cell_gain; /* left gain value in dB*/
+	u16 right_cell_gain; /* right gain value in dB*/
+	u16 gain_time_const; /* gain time constant*/
+} __packed;
+
+struct snd_sst_mute {
+	u32	stream_id;
+	u32	mute;
+};
+
+struct snd_sst_runtime_params {
+	u8 type;
+	u8 str_id;
+	u8 size;
+	u8 rsvd;
+	void *addr;
+} __packed;
+
+enum stream_param_type {
+	SST_SET_TIME_SLOT = 0,
+	SST_SET_CHANNEL_INFO = 1,
+	OTHERS = 2, /*reserved for future params*/
+};
+
+/* CSV Voice call routing structure */
+struct snd_sst_control_routing {
+	u8 control; /* 0=start, 1=Stop */
+	u8 reserved[3];	/* Reserved- for 32 bit alignment */
+};
+
+struct ipc_post {
+	struct list_head node;
+	union ipc_header header; /* driver specific */
+	bool is_large;
+	bool is_process_reply;
+	union ipc_header_mrfld mrfld_header;
+	char *mailbox_data;
+};
+
+struct snd_sst_ctxt_params {
+	u32 address; /* Physical Address in DDR where the context is stored */
+	u32 size; /* size of the context */
+};
+
+struct snd_sst_lpe_log_params {
+	u8 dbg_type;
+	u8 module_id;
+	u8 log_level;
+	u8 reserved;
+} __packed;
+
+enum snd_sst_bytes_type {
+	SND_SST_BYTES_SET = 0x1,
+	SND_SST_BYTES_GET = 0x2,
+};
+
+struct snd_sst_bytes_v2 {
+	u8 type;
+	u8 ipc_msg;
+	u8 block;
+	u8 task_id;
+	u8 pipe_id;
+	u8 rsvd;
+	u16 len;
+	char bytes[0];
+};
+
+#define MAX_VTSV_FILES 2
+struct snd_sst_vtsv_info {
+	struct sst_address_info vfiles[MAX_VTSV_FILES];
+} __packed;
+
 #endif /* __SST_MFLD_DSP_H__ */

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

@@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
 	int retval;
 	struct snd_sst_params str_params;
 	struct sst_compress_cb cb;
+	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
 
 	stream = cstream->runtime->private_data;
 	/* construct fw structure for this*/
 	memset(&str_params, 0, sizeof(str_params));
 
-	str_params.ops = STREAM_OPS_PLAYBACK;
-	str_params.stream_type = SST_STREAM_TYPE_MUSIC;
-	str_params.device_type = SND_SST_DEVICE_COMPRESS;
+	/* fill the device type and stream id to pass to SST driver */
+	retval = sst_fill_stream_params(cstream, ctx, &str_params, true);
+	pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval);
+	if (retval < 0)
+		return retval;
 
 	switch (params->codec.id) {
 	case SND_AUDIOCODEC_MP3: {

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

@@ -1,7 +1,7 @@
 /*
  *  sst_mfld_platform.c - Intel MID Platform driver
  *
- *  Copyright (C) 2010-2013 Intel Corp
+ *  Copyright (C) 2010-2014 Intel Corp
  *  Author: Vinod Koul <vinod.koul@intel.com>
  *  Author: Harsha Priya <priya.harsha@intel.com>
  *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -27,7 +27,9 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/compress_driver.h>
+#include <asm/platform_sst_audio.h>
 #include "sst-mfld-platform.h"
+#include "sst-atom-controls.h"
 
 struct sst_device *sst;
 static DEFINE_MUTEX(sst_lock);
@@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = {
 	.fifo_size = SST_FIFO_SIZE,
 };
 
+static struct sst_dev_stream_map dpcm_strm_map[] = {
+	{0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */
+	{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0},
+	{MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0},
+	{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
+};
+
 /* MFLD - MSIC */
 static struct snd_soc_dai_driver sst_platform_dai[] = {
 {
@@ -143,58 +152,142 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
 	return state;
 }
 
+static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
+				struct snd_sst_alloc_params_ext *alloc_param)
+{
+	unsigned int channels;
+	snd_pcm_uframes_t period_size;
+	ssize_t periodbytes;
+	ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+	u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+
+	channels = substream->runtime->channels;
+	period_size = substream->runtime->period_size;
+	periodbytes = samples_to_bytes(substream->runtime, period_size);
+	alloc_param->ring_buf_info[0].addr = buffer_addr;
+	alloc_param->ring_buf_info[0].size = buffer_bytes;
+	alloc_param->sg_count = 1;
+	alloc_param->reserved = 0;
+	alloc_param->frag_size = periodbytes * channels;
+
+}
 static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
-				struct sst_pcm_params *param)
+				struct snd_sst_stream_params *param)
 {
+	param->uc.pcm_params.num_chan = (u8) substream->runtime->channels;
+	param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits;
+	param->uc.pcm_params.sfreq = substream->runtime->rate;
+
+	/* PCM stream via ALSA interface */
+	param->uc.pcm_params.use_offload_path = 0;
+	param->uc.pcm_params.reserved2 = 0;
+	memset(param->uc.pcm_params.channel_map, 0, sizeof(u8));
 
-	param->num_chan = (u8) substream->runtime->channels;
-	param->pcm_wd_sz = substream->runtime->sample_bits;
-	param->reserved = 0;
-	param->sfreq = substream->runtime->rate;
-	param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream);
-	param->period_count = substream->runtime->period_size;
-	param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area);
-	pr_debug("period_cnt = %d\n", param->period_count);
-	pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz);
 }
 
-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
+static int sst_get_stream_mapping(int dev, int sdev, int dir,
+	struct sst_dev_stream_map *map, int size)
+{
+	int i;
+
+	if (map == NULL)
+		return -EINVAL;
+
+
+	/* index 0 is not used in stream map */
+	for (i = 1; i < size; i++) {
+		if ((map[i].dev_num == dev) && (map[i].direction == dir))
+			return i;
+	}
+	return 0;
+}
+
+int sst_fill_stream_params(void *substream,
+	const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress)
+{
+	int map_size;
+	int index;
+	struct sst_dev_stream_map *map;
+	struct snd_pcm_substream *pstream = NULL;
+	struct snd_compr_stream *cstream = NULL;
+
+	map = ctx->pdata->pdev_strm_map;
+	map_size = ctx->pdata->strm_map_size;
+
+	if (is_compress == true)
+		cstream = (struct snd_compr_stream *)substream;
+	else
+		pstream = (struct snd_pcm_substream *)substream;
+
+	str_params->stream_type = SST_STREAM_TYPE_MUSIC;
+
+	/* For pcm streams */
+	if (pstream) {
+		index = sst_get_stream_mapping(pstream->pcm->device,
+					  pstream->number, pstream->stream,
+					  map, map_size);
+		if (index <= 0)
+			return -EINVAL;
+
+		str_params->stream_id = index;
+		str_params->device_type = map[index].device_id;
+		str_params->task = map[index].task_id;
+
+		str_params->ops = (u8)pstream->stream;
+	}
+
+	if (cstream) {
+		index = sst_get_stream_mapping(cstream->device->device,
+					       0, cstream->direction,
+					       map, map_size);
+		if (index <= 0)
+			return -EINVAL;
+		str_params->stream_id = index;
+		str_params->device_type = map[index].device_id;
+		str_params->task = map[index].task_id;
+
+		str_params->ops = (u8)cstream->direction;
+	}
+	return 0;
+}
+
+static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
+		struct snd_soc_platform *platform)
 {
 	struct sst_runtime_stream *stream =
 			substream->runtime->private_data;
-	struct sst_pcm_params param = {0};
-	struct sst_stream_params str_params = {0};
-	int ret_val;
+	struct snd_sst_stream_params param = {{{0,},},};
+	struct snd_sst_params str_params = {0};
+	struct snd_sst_alloc_params_ext alloc_params = {0};
+	int ret_val = 0;
+	struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
 
 	/* set codec params and inform SST driver the same */
 	sst_fill_pcm_params(substream, &param);
+	sst_fill_alloc_params(substream, &alloc_params);
 	substream->runtime->dma_area = substream->dma_buffer.area;
 	str_params.sparams = param;
-	str_params.codec =  param.codec;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		str_params.ops = STREAM_OPS_PLAYBACK;
-		str_params.device_type = substream->pcm->device + 1;
-		pr_debug("Playbck stream,Device %d\n",
-					substream->pcm->device);
-	} else {
-		str_params.ops = STREAM_OPS_CAPTURE;
-		str_params.device_type = SND_SST_DEVICE_CAPTURE;
-		pr_debug("Capture stream,Device %d\n",
-					substream->pcm->device);
-	}
-	ret_val = stream->ops->open(&str_params);
-	pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
+	str_params.aparams = alloc_params;
+	str_params.codec = SST_CODEC_TYPE_PCM;
+
+	/* fill the device type and stream id to pass to SST driver */
+	ret_val = sst_fill_stream_params(substream, ctx, &str_params, false);
 	if (ret_val < 0)
 		return ret_val;
 
-	stream->stream_info.str_id = ret_val;
-	pr_debug("str id :  %d\n", stream->stream_info.str_id);
+	stream->stream_info.str_id = str_params.stream_id;
+
+	ret_val = stream->ops->open(&str_params);
+	if (ret_val <= 0)
+		return ret_val;
+
+
 	return ret_val;
 }
 
-static void sst_period_elapsed(void *mad_substream)
+static void sst_period_elapsed(void *arg)
 {
-	struct snd_pcm_substream *substream = mad_substream;
+	struct snd_pcm_substream *substream = arg;
 	struct sst_runtime_stream *stream;
 	int status;
 
@@ -218,7 +311,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 	pr_debug("setting buffer ptr param\n");
 	sst_set_stream_status(stream, SST_PLATFORM_INIT);
 	stream->stream_info.period_elapsed = sst_period_elapsed;
-	stream->stream_info.mad_substream = substream;
+	stream->stream_info.arg = substream;
 	stream->stream_info.buffer_ptr = 0;
 	stream->stream_info.sfreq = substream->runtime->rate;
 	ret_val = stream->ops->device_control(
@@ -230,19 +323,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 }
 /* end -- helper functions */
 
-static int sst_platform_open(struct snd_pcm_substream *substream)
+static int sst_media_open(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
 {
+	int ret_val = 0;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct sst_runtime_stream *stream;
-	int ret_val;
-
-	pr_debug("sst_platform_open called\n");
-
-	snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
-	ret_val = snd_pcm_hw_constraint_integer(runtime,
-						SNDRV_PCM_HW_PARAM_PERIODS);
-	if (ret_val < 0)
-		return ret_val;
 
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
@@ -251,50 +337,69 @@ static int sst_platform_open(struct snd_pcm_substream *substream)
 
 	/* get the sst ops */
 	mutex_lock(&sst_lock);
-	if (!sst) {
+	if (!sst ||
+	    !try_module_get(sst->dev->driver->owner)) {
 		pr_err("no device available to run\n");
-		mutex_unlock(&sst_lock);
-		kfree(stream);
-		return -ENODEV;
-	}
-	if (!try_module_get(sst->dev->driver->owner)) {
-		mutex_unlock(&sst_lock);
-		kfree(stream);
-		return -ENODEV;
+		ret_val = -ENODEV;
+		goto out_ops;
 	}
 	stream->ops = sst->ops;
 	mutex_unlock(&sst_lock);
 
 	stream->stream_info.str_id = 0;
-	sst_set_stream_status(stream, SST_PLATFORM_INIT);
-	stream->stream_info.mad_substream = substream;
+
+	stream->stream_info.arg = substream;
 	/* allocate memory for SST API set */
 	runtime->private_data = stream;
 
-	return 0;
+	/* Make sure, that the period size is always even */
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_PERIODS, 2);
+
+	return snd_pcm_hw_constraint_integer(runtime,
+			 SNDRV_PCM_HW_PARAM_PERIODS);
+out_ops:
+	kfree(stream);
+	mutex_unlock(&sst_lock);
+	return ret_val;
 }
 
-static int sst_platform_close(struct snd_pcm_substream *substream)
+static void sst_media_close(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
 {
 	struct sst_runtime_stream *stream;
 	int ret_val = 0, str_id;
 
-	pr_debug("sst_platform_close called\n");
 	stream = substream->runtime->private_data;
 	str_id = stream->stream_info.str_id;
 	if (str_id)
 		ret_val = stream->ops->close(str_id);
 	module_put(sst->dev->driver->owner);
 	kfree(stream);
-	return ret_val;
 }
 
-static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
+static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform,
+					       struct snd_pcm_substream *substream)
+{
+	struct sst_data *sst = snd_soc_platform_get_drvdata(platform);
+	struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	u32 str_id = stream->stream_info.str_id;
+	unsigned int pipe_id;
+	pipe_id = map[str_id].device_id;
+
+	pr_debug("%s: got pipe_id = %#x for str_id = %d\n",
+		 __func__, pipe_id, str_id);
+	return pipe_id;
+}
+
+static int sst_media_prepare(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
 {
 	struct sst_runtime_stream *stream;
 	int ret_val = 0, str_id;
 
-	pr_debug("sst_platform_pcm_prepare called\n");
 	stream = substream->runtime->private_data;
 	str_id = stream->stream_info.str_id;
 	if (stream->stream_info.str_id) {
@@ -303,8 +408,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
 		return ret_val;
 	}
 
-	ret_val = sst_platform_alloc_stream(substream);
-	if (ret_val < 0)
+	ret_val = sst_platform_alloc_stream(substream, dai->platform);
+	if (ret_val <= 0)
 		return ret_val;
 	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
 			"%d", stream->stream_info.str_id);
@@ -316,6 +421,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
 	return ret_val;
 }
 
+static int sst_media_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+	return 0;
+}
+
+static int sst_media_hw_free(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_soc_dai_ops sst_media_dai_ops = {
+	.startup = sst_media_open,
+	.shutdown = sst_media_close,
+	.prepare = sst_media_prepare,
+	.hw_params = sst_media_hw_params,
+	.hw_free = sst_media_hw_free,
+};
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime;
+
+	if (substream->pcm->internal)
+		return 0;
+
+	runtime = substream->runtime;
+	runtime->hw = sst_platform_pcm_hw;
+	return 0;
+}
+
 static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
 					int cmd)
 {
@@ -331,7 +471,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
 		pr_debug("sst: Trigger Start\n");
 		str_cmd = SST_SND_START;
 		status = SST_PLATFORM_RUNNING;
-		stream->stream_info.mad_substream = substream;
+		stream->stream_info.arg = substream;
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		pr_debug("sst: in stop\n");
@@ -377,32 +517,15 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer
 		pr_err("sst: error code = %d\n", ret_val);
 		return ret_val;
 	}
-	return stream->stream_info.buffer_ptr;
-}
-
-static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params)
-{
-	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
-	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-
-	return 0;
-}
-
-static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	return snd_pcm_lib_free_pages(substream);
+	substream->runtime->delay = str_info->pcm_delay;
+	return str_info->buffer_ptr;
 }
 
 static struct snd_pcm_ops sst_platform_ops = {
 	.open = sst_platform_open,
-	.close = sst_platform_close,
 	.ioctl = snd_pcm_lib_ioctl,
-	.prepare = sst_platform_pcm_prepare,
 	.trigger = sst_platform_pcm_trigger,
 	.pointer = sst_platform_pcm_pointer,
-	.hw_params = sst_platform_pcm_hw_params,
-	.hw_free = sst_platform_pcm_hw_free,
 };
 
 static void sst_pcm_free(struct snd_pcm *pcm)
@@ -413,15 +536,15 @@ static void sst_pcm_free(struct snd_pcm *pcm)
 
 static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int retval = 0;
 
-	pr_debug("sst_pcm_new called\n");
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
-			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+	if (dai->driver->playback.channels_min ||
+			dai->driver->capture.channels_min) {
 		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
 			SNDRV_DMA_TYPE_CONTINUOUS,
-			snd_dma_continuous_data(GFP_KERNEL),
+			snd_dma_continuous_data(GFP_DMA),
 			SST_MIN_BUFFER, SST_MAX_BUFFER);
 		if (retval) {
 			pr_err("dma buffer allocationf fail\n");
@@ -445,10 +568,28 @@ static const struct snd_soc_component_driver sst_component = {
 
 static int sst_platform_probe(struct platform_device *pdev)
 {
+	struct sst_data *drv;
 	int ret;
+	struct sst_platform_data *pdata;
+
+	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
+	if (drv == NULL) {
+		pr_err("kzalloc failed\n");
+		return -ENOMEM;
+	}
+
+	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+	if (pdata == NULL) {
+		pr_err("kzalloc failed for pdata\n");
+		return -ENOMEM;
+	}
+
+	pdata->pdev_strm_map = dpcm_strm_map;
+	pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map);
+	drv->pdata = pdata;
+	mutex_init(&drv->lock);
+	dev_set_drvdata(&pdev->dev, drv);
 
-	pr_debug("sst_platform_probe called\n");
-	sst = NULL;
 	ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
 	if (ret) {
 		pr_err("registering soc platform failed\n");

sound/soc/intel/sst-mfld-platform.h

@@ -39,9 +39,10 @@ extern struct sst_device *sst;
 
 struct pcm_stream_info {
 	int str_id;
-	void *mad_substream;
-	void (*period_elapsed) (void *mad_substream);
+	void *arg;
+	void (*period_elapsed) (void *arg);
 	unsigned long long buffer_ptr;
+	unsigned long long pcm_delay;
 	int sfreq;
 };
 
@@ -62,7 +63,9 @@ enum sst_controls {
 	SST_SND_BUFFER_POINTER =	0x05,
 	SST_SND_STREAM_INIT =		0x06,
 	SST_SND_START	 =		0x07,
-	SST_MAX_CONTROLS =		0x07,
+	SST_SET_BYTE_STREAM =           0x100A,
+	SST_GET_BYTE_STREAM =           0x100B,
+	SST_MAX_CONTROLS = SST_GET_BYTE_STREAM,
 };
 
 enum sst_stream_ops {
@@ -124,8 +127,9 @@ struct compress_sst_ops {
 };
 
 struct sst_ops {
-	int (*open) (struct sst_stream_params *str_param);
+	int (*open) (struct snd_sst_params *str_param);
 	int (*device_control) (int cmd, void *arg);
+	int (*set_generic_params)(enum sst_controls cmd, void *arg);
 	int (*close) (unsigned int str_id);
 };
 
@@ -143,10 +147,27 @@ struct sst_device {
 	char *name;
 	struct device *dev;
 	struct sst_ops *ops;
+	struct platform_device *pdev;
 	struct compress_sst_ops *compr_ops;
 };
 
+struct sst_data;
 void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
+int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
+			   struct snd_sst_params *str_params, bool is_compress);
+
+struct sst_algo_int_control_v2 {
+	struct soc_mixer_control mc;
+	u16 module_id; /* module identifieer */
+	u16 pipe_id; /* location info: pipe_id + instance_id */
+	u16 instance_id;
+	unsigned int value; /* Value received is stored here */
+};
+struct sst_data {
+	struct platform_device *pdev;
+	struct sst_platform_data *pdata;
+	struct mutex lock;
+};
 int sst_register_dsp(struct sst_device *sst);
 int sst_unregister_dsp(struct sst_device *sst);
 #endif

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 || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST
+	depends on 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
@@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB
 	  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)
-	depends on I2C
-	select SND_SOC_CS42L51
-	help
-	  Say Y if you want to add support for SoC audio on
-	  Openrd Client.
-
-config SND_KIRKWOOD_SOC_T5325
-	tristate "SoC Audio support for HP t5325"
-	depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C
-	select SND_SOC_ALC5623
-	help
-	  Say Y if you want to add support for SoC audio on
-	  the HP t5325 thin client.
-

sound/soc/kirkwood/Makefile

@@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o
 
 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

sound/soc/kirkwood/kirkwood-dma.c

@@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
 }
 
 static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
-	.info = (SNDRV_PCM_INFO_INTERLEAVED |
-		 SNDRV_PCM_INFO_MMAP |
-		 SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_PAUSE),
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
 	.buffer_bytes_max	= KIRKWOOD_SND_MAX_BUFFER_BYTES,
 	.period_bytes_min	= KIRKWOOD_SND_MIN_PERIOD_BYTES,
 	.period_bytes_max	= KIRKWOOD_SND_MAX_PERIOD_BYTES,

sound/soc/kirkwood/kirkwood-i2s.c

@@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
 				    KIRKWOOD_PLAYCTL_SIZE_MASK);
 		priv->ctl_play |= ctl_play;
 	} else {
-		priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK;
+		priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK |
+				   KIRKWOOD_RECCTL_SIZE_MASK);
 		priv->ctl_rec |= ctl_rec;
 	}
 
@@ -221,14 +222,24 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static unsigned kirkwood_i2s_play_mute(unsigned ctl)
+{
+	if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN))
+		ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE;
+	if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN))
+		ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE;
+	return ctl;
+}
+
 static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 				int cmd, struct snd_soc_dai *dai)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai);
 	uint32_t ctl, value;
 
 	ctl = readl(priv->io + KIRKWOOD_PLAYCTL);
-	if (ctl & KIRKWOOD_PLAYCTL_PAUSE) {
+	if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) {
 		unsigned timeout = 5000;
 		/*
 		 * The Armada510 spec says that if we enter pause mode, the
@@ -256,14 +267,16 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 			ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;	/* i2s */
 		else
 			ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;	/* spdif */
-
+		ctl = kirkwood_i2s_play_mute(ctl);
 		value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
 		/* enable interrupts */
-		value = readl(priv->io + KIRKWOOD_INT_MASK);
-		value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
-		writel(value, priv->io + KIRKWOOD_INT_MASK);
+		if (!runtime->no_period_wakeup) {
+			value = readl(priv->io + KIRKWOOD_INT_MASK);
+			value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
+			writel(value, priv->io + KIRKWOOD_INT_MASK);
+		}
 
 		/* enable playback */
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
@@ -295,6 +308,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
 				KIRKWOOD_PLAYCTL_SPDIF_MUTE);
+		ctl = kirkwood_i2s_play_mute(ctl);
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
 		break;
 
@@ -322,8 +336,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
 		else
 			ctl &= ~KIRKWOOD_RECCTL_I2S_EN;		/* spdif */
 
-		value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN |
-				KIRKWOOD_RECCTL_SPDIF_EN);
+		value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_RECCTL);
 
 		/* enable interrupts */
@@ -347,7 +360,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
 
 		/* disable all records */
 		value = readl(priv->io + KIRKWOOD_RECCTL);
-		value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+		value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_RECCTL);
 		break;
 
@@ -411,7 +424,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv)
 	writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
 	value = readl(priv->io + KIRKWOOD_RECCTL);
-	value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+	value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
 	writel(value, priv->io + KIRKWOOD_RECCTL);
 
 	return 0;

sound/soc/kirkwood/kirkwood-openrd.c

@@ -1,109 +0,0 @@
-/*
- * kirkwood-openrd.c
- *
- * (c) 2010 Arnaud Patard <apatard@mandriva.com>
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- *  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/platform_data/asoc-kirkwood.h>
-#include "../codecs/cs42l51.h"
-
-static int openrd_client_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 openrd_client_ops = {
-	.hw_params = openrd_client_hw_params,
-};
-
-
-static struct snd_soc_dai_link openrd_client_dai[] = {
-{
-	.name = "CS42L51",
-	.stream_name = "CS42L51 HiFi",
-	.cpu_dai_name = "i2s",
-	.platform_name = "mvebu-audio",
-	.codec_dai_name = "cs42l51-hifi",
-	.codec_name = "cs42l51-codec.0-004a",
-	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
-	.ops = &openrd_client_ops,
-},
-};
-
-
-static struct snd_soc_card openrd_client = {
-	.name = "OpenRD Client",
-	.owner = THIS_MODULE,
-	.dai_link = openrd_client_dai,
-	.num_links = ARRAY_SIZE(openrd_client_dai),
-};
-
-static int openrd_probe(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = &openrd_client;
-	int ret;
-
-	card->dev = &pdev->dev;
-
-	ret = snd_soc_register_card(card);
-	if (ret)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
-	return ret;
-}
-
-static int openrd_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
-static struct platform_driver openrd_driver = {
-	.driver		= {
-		.name	= "openrd-client-audio",
-		.owner	= THIS_MODULE,
-	},
-	.probe		= openrd_probe,
-	.remove		= openrd_remove,
-};
-
-module_platform_driver(openrd_driver);
-
-/* Module information */
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC OpenRD Client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:openrd-client-audio");

sound/soc/kirkwood/kirkwood-t5325.c

@@ -1,116 +0,0 @@
-/*
- * kirkwood-t5325.c
- *
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- *  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/platform_data/asoc-kirkwood.h>
-#include "../codecs/alc5623.h"
-
-static int t5325_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;
-
-	freq = params_rate(params) * 256;
-
-	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops t5325_ops = {
-	.hw_params = t5325_hw_params,
-};
-
-static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone Jack", NULL),
-	SND_SOC_DAPM_SPK("Speaker", NULL),
-	SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route t5325_route[] = {
-	{ "Headphone Jack",	NULL,	"HPL" },
-	{ "Headphone Jack",	NULL,	"HPR" },
-
-	{"Speaker",		NULL,	"SPKOUT"},
-	{"Speaker",		NULL,	"SPKOUTN"},
-
-	{ "MIC1",		NULL,	"Mic Jack" },
-	{ "MIC2",		NULL,	"Mic Jack" },
-};
-
-static struct snd_soc_dai_link t5325_dai[] = {
-{
-	.name = "ALC5621",
-	.stream_name = "ALC5621 HiFi",
-	.cpu_dai_name = "i2s",
-	.platform_name = "mvebu-audio",
-	.codec_dai_name = "alc5621-hifi",
-	.codec_name = "alc562x-codec.0-001a",
-	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
-	.ops = &t5325_ops,
-},
-};
-
-static struct snd_soc_card t5325 = {
-	.name = "t5325",
-	.owner = THIS_MODULE,
-	.dai_link = t5325_dai,
-	.num_links = ARRAY_SIZE(t5325_dai),
-
-	.dapm_widgets = t5325_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets),
-	.dapm_routes = t5325_route,
-	.num_dapm_routes = ARRAY_SIZE(t5325_route),
-};
-
-static int t5325_probe(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = &t5325;
-	int ret;
-
-	card->dev = &pdev->dev;
-
-	ret = snd_soc_register_card(card);
-	if (ret)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
-	return ret;
-}
-
-static int t5325_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
-static struct platform_driver t5325_driver = {
-	.driver		= {
-		.name	= "t5325-audio",
-		.owner	= THIS_MODULE,
-	},
-	.probe		= t5325_probe,
-	.remove		= t5325_remove,
-};
-
-module_platform_driver(t5325_driver);
-
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:t5325-audio");

sound/soc/kirkwood/kirkwood.h

@@ -38,6 +38,9 @@
 #define KIRKWOOD_RECCTL_SIZE_24		(1<<0)
 #define KIRKWOOD_RECCTL_SIZE_32		(0<<0)
 
+#define KIRKWOOD_RECCTL_ENABLE_MASK		(KIRKWOOD_RECCTL_SPDIF_EN | \
+						 KIRKWOOD_RECCTL_I2S_EN)
+
 #define KIRKWOOD_REC_BUF_ADDR			0x1004
 #define KIRKWOOD_REC_BUF_SIZE			0x1008
 #define KIRKWOOD_REC_BYTE_COUNT			0x100C
@@ -121,9 +124,9 @@
 
 /* Theses values come from the marvell alsa driver */
 /* need to find where they come from               */
-#define KIRKWOOD_SND_MIN_PERIODS		8
+#define KIRKWOOD_SND_MIN_PERIODS		2
 #define KIRKWOOD_SND_MAX_PERIODS		16
-#define KIRKWOOD_SND_MIN_PERIOD_BYTES		0x800
+#define KIRKWOOD_SND_MIN_PERIOD_BYTES		256
 #define KIRKWOOD_SND_MAX_PERIOD_BYTES		0x8000
 #define KIRKWOOD_SND_MAX_BUFFER_BYTES		(KIRKWOOD_SND_MAX_PERIOD_BYTES \
 						 * KIRKWOOD_SND_MAX_PERIODS)