Add the rt linux 4.1.3-rt3 as base

Import the rt linux 4.1.3-rt3 as OPNFV kvm base.

It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and
the base is:

commit 0917f823c59692d751951bf5ea699a2d1e2f26a2
Author: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Date:   Sat Jul 25 12:13:34 2015 +0200

    Prepare v4.1.3-rt3

    Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>

We lose all the git history this way and it's not good. We
should apply another opnfv project repo in future.

Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423
Signed-off-by: Yunhong Jiang <yunhong.jiang@intel.com>

9 files changed, 2279 insertions, 0 deletions

diff --git a/kernel/sound/soc/intel/boards/Makefile b/kernel/sound/soc/intel/boards/Makefile
new file mode 100644
index 000000000..f8237f004
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/Makefile
@@ -0,0 +1,15 @@
+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
+snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
+snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
+snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
+
+obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
diff --git a/kernel/sound/soc/intel/boards/broadwell.c b/kernel/sound/soc/intel/boards/broadwell.c
new file mode 100644
index 000000000..8bafaf6ce
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/broadwell.c
@@ -0,0 +1,292 @@
+/*
+ * 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/jack.h>
+#include <sound/pcm_params.h>
+
+#include "../common/sst-dsp.h"
+#include "../haswell/sst-haswell-ipc.h"
+
+#include "../../codecs/rt286.h"
+
+static struct snd_soc_jack broadwell_headset;
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin broadwell_headset_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headphone Jack",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static const struct snd_kcontrol_new broadwell_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+};
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", 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 */
+	{"Headphone Jack", 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_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret = 0;
+	ret = snd_soc_card_jack_new(rtd->card, "Headset",
+		SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset,
+		broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins));
+	if (ret)
+		return ret;
+
+	rt286_mic_detect(codec, &broadwell_headset);
+	return 0;
+}
+
+
+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 */
+	params_set_format(params, 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 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;
+	}
+
+	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/Capture",
+		.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,
+		.dpcm_capture = 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,
+	},
+	/* 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",
+		.init = broadwell_rt286_codec_init,
+		.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,
+	},
+};
+
+static int broadwell_suspend(struct snd_soc_card *card){
+	struct snd_soc_codec *codec;
+
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+		if (!strcmp(codec->component.name, "i2c-INT343A:00")) {
+			dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
+			rt286_mic_detect(codec, NULL);
+			break;
+		}
+	}
+	return 0;
+}
+
+static int broadwell_resume(struct snd_soc_card *card){
+	struct snd_soc_codec *codec;
+
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+		if (!strcmp(codec->component.name, "i2c-INT343A:00")) {
+			dev_dbg(codec->dev, "enabling jack detect for resume.\n");
+			rt286_mic_detect(codec, &broadwell_headset);
+			break;
+		}
+	}
+	return 0;
+}
+
+/* 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),
+	.controls = broadwell_controls,
+	.num_controls = ARRAY_SIZE(broadwell_controls),
+	.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,
+	.suspend_pre = broadwell_suspend,
+	.resume_post = broadwell_resume,
+};
+
+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",
+	},
+};
+
+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");
diff --git a/kernel/sound/soc/intel/boards/byt-max98090.c b/kernel/sound/soc/intel/boards/byt-max98090.c
new file mode 100644
index 000000000..7ab8cc9fb
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/byt-max98090.c
@@ -0,0 +1,187 @@
+/*
+ * Intel Baytrail SST MAX98090 machine driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/max98090.h"
+
+struct byt_max98090_private {
+	struct snd_soc_jack jack;
+};
+
+static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
+	{"IN34", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "MICBIAS"},
+	{"DMICL", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPL"},
+	{"Headphone", NULL, "HPR"},
+	{"Ext Spk", NULL, "SPKL"},
+	{"Ext Spk", NULL, "SPKR"},
+};
+
+static const struct snd_kcontrol_new byt_max98090_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin	= "Headphone",
+		.mask	= SND_JACK_HEADPHONE,
+	},
+	{
+		.pin	= "Headset Mic",
+		.mask	= SND_JACK_MICROPHONE,
+	},
+};
+
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.name		= "hp-gpio",
+		.idx		= 0,
+		.report		= SND_JACK_HEADPHONE | SND_JACK_LINEOUT,
+		.debounce_time	= 200,
+	},
+	{
+		.name		= "mic-gpio",
+		.idx		= 1,
+		.invert		= 1,
+		.report		= SND_JACK_MICROPHONE,
+		.debounce_time	= 200,
+	},
+};
+
+static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_card *card = runtime->card;
+	struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_jack *jack = &drv->jack;
+
+	card->dapm.idle_bias_off = true;
+
+	ret = snd_soc_dai_set_sysclk(runtime->codec_dai,
+				     M98090_REG_SYSTEM_CLOCK,
+				     25000000, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "Can't set codec clock %d\n", ret);
+		return ret;
+	}
+
+	/* Enable jack detection */
+	ret = snd_soc_card_jack_new(runtime->card, "Headset",
+				    SND_JACK_LINEOUT | SND_JACK_HEADSET, jack,
+				    hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
+	if (ret)
+		return ret;
+
+	return snd_soc_jack_add_gpiods(card->dev->parent, jack,
+				       ARRAY_SIZE(hs_jack_gpios),
+				       hs_jack_gpios);
+}
+
+static struct snd_soc_dai_link byt_max98090_dais[] = {
+	{
+		.name = "Baytrail Audio",
+		.stream_name = "Audio",
+		.cpu_dai_name = "baytrail-pcm-audio",
+		.codec_dai_name = "HiFi",
+		.codec_name = "i2c-193C9890:00",
+		.platform_name = "baytrail-pcm-audio",
+		.init = byt_max98090_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+	},
+};
+
+static struct snd_soc_card byt_max98090_card = {
+	.name = "byt-max98090",
+	.dai_link = byt_max98090_dais,
+	.num_links = ARRAY_SIZE(byt_max98090_dais),
+	.dapm_widgets = byt_max98090_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets),
+	.dapm_routes = byt_max98090_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map),
+	.controls = byt_max98090_controls,
+	.num_controls = ARRAY_SIZE(byt_max98090_controls),
+	.fully_routed = true,
+};
+
+static int byt_max98090_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	struct byt_max98090_private *priv;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC);
+	if (!priv) {
+		dev_err(&pdev->dev, "allocation failed\n");
+		return -ENOMEM;
+	}
+
+	byt_max98090_card.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&byt_max98090_card, priv);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_max98090_card);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+
+	return ret_val;
+}
+
+static int byt_max98090_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card);
+
+	snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
+	return 0;
+}
+
+static struct platform_driver byt_max98090_driver = {
+	.probe = byt_max98090_probe,
+	.remove = byt_max98090_remove,
+	.driver = {
+		.name = "byt-max98090",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+module_platform_driver(byt_max98090_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver");
+MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:byt-max98090");
diff --git a/kernel/sound/soc/intel/boards/byt-rt5640.c b/kernel/sound/soc/intel/boards/byt-rt5640.c
new file mode 100644
index 000000000..ae89b9b96
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/byt-rt5640.c
@@ -0,0 +1,229 @@
+/*
+ * Intel Baytrail SST RT5640 machine driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/dmi.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5640.h"
+
+#include "../common/sst-dsp.h"
+
+static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Internal Mic", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+	{"Headset Mic", NULL, "MICBIAS1"},
+	{"IN2P", NULL, "Headset Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Speaker", NULL, "SPOLP"},
+	{"Speaker", NULL, "SPOLN"},
+	{"Speaker", NULL, "SPORP"},
+	{"Speaker", NULL, "SPORN"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = {
+	{"DMIC1", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = {
+	{"DMIC2", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = {
+	{"Internal Mic", NULL, "MICBIAS1"},
+	{"IN1P", NULL, "Internal Mic"},
+};
+
+enum {
+	BYT_RT5640_DMIC1_MAP,
+	BYT_RT5640_DMIC2_MAP,
+	BYT_RT5640_IN1_MAP,
+};
+
+#define BYT_RT5640_MAP(quirk)	((quirk) & 0xff)
+#define BYT_RT5640_DMIC_EN	BIT(16)
+
+static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_DMIC_EN;
+
+static const struct snd_kcontrol_new byt_rt5640_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Internal Mic"),
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int byt_rt5640_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, RT5640_SCLK_S_PLL1,
+				     params_rate(params) * 256,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "can't set codec clock %d\n", ret);
+		return ret;
+	}
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+				  params_rate(params) * 64,
+				  params_rate(params) * 256);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+	return 0;
+}
+
+static int byt_rt5640_quirk_cb(const struct dmi_system_id *id)
+{
+	byt_rt5640_quirk = (unsigned long)id->driver_data;
+	return 1;
+}
+
+static const struct dmi_system_id byt_rt5640_quirk_table[] = {
+	{
+		.callback = byt_rt5640_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"),
+		},
+		.driver_data = (unsigned long *)BYT_RT5640_IN1_MAP,
+	},
+	{
+		.callback = byt_rt5640_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "DellInc."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"),
+		},
+		.driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP |
+						 BYT_RT5640_DMIC_EN),
+	},
+	{}
+};
+
+static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_card *card = runtime->card;
+	const struct snd_soc_dapm_route *custom_map;
+	int num_routes;
+
+	card->dapm.idle_bias_off = true;
+
+	ret = snd_soc_add_card_controls(card, byt_rt5640_controls,
+					ARRAY_SIZE(byt_rt5640_controls));
+	if (ret) {
+		dev_err(card->dev, "unable to add card controls\n");
+		return ret;
+	}
+
+	dmi_check_system(byt_rt5640_quirk_table);
+	switch (BYT_RT5640_MAP(byt_rt5640_quirk)) {
+	case BYT_RT5640_IN1_MAP:
+		custom_map = byt_rt5640_intmic_in1_map;
+		num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map);
+		break;
+	case BYT_RT5640_DMIC2_MAP:
+		custom_map = byt_rt5640_intmic_dmic2_map;
+		num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map);
+		break;
+	default:
+		custom_map = byt_rt5640_intmic_dmic1_map;
+		num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map);
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes);
+	if (ret)
+		return ret;
+
+	if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) {
+		ret = rt5640_dmic_enable(codec, 0, 0);
+		if (ret)
+			return ret;
+	}
+
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+
+	return ret;
+}
+
+static struct snd_soc_ops byt_rt5640_ops = {
+	.hw_params = byt_rt5640_hw_params,
+};
+
+static struct snd_soc_dai_link byt_rt5640_dais[] = {
+	{
+		.name = "Baytrail Audio",
+		.stream_name = "Audio",
+		.cpu_dai_name = "baytrail-pcm-audio",
+		.codec_dai_name = "rt5640-aif1",
+		.codec_name = "i2c-10EC5640:00",
+		.platform_name = "baytrail-pcm-audio",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.init = byt_rt5640_init,
+		.ops = &byt_rt5640_ops,
+	},
+};
+
+static struct snd_soc_card byt_rt5640_card = {
+	.name = "byt-rt5640",
+	.dai_link = byt_rt5640_dais,
+	.num_links = ARRAY_SIZE(byt_rt5640_dais),
+	.dapm_widgets = byt_rt5640_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets),
+	.dapm_routes = byt_rt5640_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map),
+	.fully_routed = true,
+};
+
+static int byt_rt5640_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &byt_rt5640_card;
+
+	card->dev = &pdev->dev;
+	return devm_snd_soc_register_card(&pdev->dev, card);
+}
+
+static struct platform_driver byt_rt5640_audio = {
+	.probe = byt_rt5640_probe,
+	.driver = {
+		.name = "byt-rt5640",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+module_platform_driver(byt_rt5640_audio)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver");
+MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:byt-rt5640");
diff --git a/kernel/sound/soc/intel/boards/bytcr_rt5640.c b/kernel/sound/soc/intel/boards/bytcr_rt5640.c
new file mode 100644
index 000000000..7f55d5902
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/bytcr_rt5640.c
@@ -0,0 +1,227 @@
+/*
+ *  byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform
+ *
+ *  Copyright (C) 2014 Intel Corp
+ *  Author: Subhransu S. Prusty <subhransu.s.prusty@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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/input.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/rt5640.h"
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget byt_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_audio_map[] = {
+	{"IN2P", NULL, "Headset Mic"},
+	{"IN2N", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "MICBIAS1"},
+	{"IN1P", NULL, "MICBIAS1"},
+	{"LDO2", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOLP"},
+	{"Ext Spk", NULL, "SPOLN"},
+	{"Ext Spk", NULL, "SPORP"},
+	{"Ext Spk", NULL, "SPORN"},
+
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+};
+
+static const struct snd_kcontrol_new byt_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int byt_aif1_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	snd_soc_dai_set_bclk_ratio(codec_dai, 50);
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+				     params_rate(params) * 512,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec clock %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+				  params_rate(params) * 50,
+				  params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_pcm_stream byt_dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	.rate_min = 48000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int byt_aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops byt_aif1_ops = {
+	.startup = byt_aif1_startup,
+};
+
+static struct snd_soc_ops byt_be_ssp2_ops = {
+	.hw_params = byt_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link byt_dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Baytrail Audio Port",
+		.stream_name = "Baytrail Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &byt_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Baytrail Compressed Port",
+		.stream_name = "Baytrail Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+		/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5640-aif1",
+		.codec_name = "i2c-10EC5640:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = byt_codec_fixup,
+		.ignore_suspend = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &byt_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_byt = {
+	.name = "baytrailcraudio",
+	.dai_link = byt_dailink,
+	.num_links = ARRAY_SIZE(byt_dailink),
+	.dapm_widgets = byt_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets),
+	.dapm_routes = byt_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(byt_audio_map),
+	.controls = byt_mc_controls,
+	.num_controls = ARRAY_SIZE(byt_mc_controls),
+};
+
+static int snd_byt_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+
+	/* register the soc card */
+	snd_soc_card_byt.dev = &pdev->dev;
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt);
+	if (ret_val) {
+		dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_byt);
+	return ret_val;
+}
+
+static struct platform_driver snd_byt_mc_driver = {
+	.driver = {
+		.name = "bytt100_rt5640",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = snd_byt_mc_probe,
+};
+
+module_platform_driver(snd_byt_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytt100_rt5640");
diff --git a/kernel/sound/soc/intel/boards/cht_bsw_rt5645.c b/kernel/sound/soc/intel/boards/cht_bsw_rt5645.c
new file mode 100644
index 000000000..20a28b22e
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -0,0 +1,324 @@
+/*
+ *  cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
+ *                     Cherrytrail and Braswell, with RT5645 codec.
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  Author: Fang, Yang A <yang.a.fang@intel.com>
+ *	        N,Harshapriya <harshapriya.n@intel.com>
+ *  This file is modified from cht_bsw_rt5672.c
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5645.h"
+#include "../atom/sst-atom-controls.h"
+
+#define CHT_PLAT_CLK_3_HZ	19200000
+#define CHT_CODEC_DAI	"rt5645-aif1"
+
+struct cht_mc_private {
+	struct snd_soc_jack hp_jack;
+	struct snd_soc_jack mic_jack;
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+	int i;
+
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_pcm_runtime *rtd;
+
+		rtd = card->rtd + i;
+		if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+			     strlen(CHT_CODEC_DAI)))
+			return rtd->codec_dai;
+	}
+	return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret;
+
+	codec_dai = cht_get_codec_dai(card);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+		return -EIO;
+	}
+
+	if (!SND_SOC_DAPM_EVENT_OFF(event))
+		return 0;
+
+	/* Set codec sysclk source to its internal clock because codec PLL will
+	 * be off when idle and MCLK will also be off by ACPI when codec is
+	 * runtime suspended. Codec needs clock for jack detection and button
+	 * press.
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
+			0, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"DMIC L1", NULL, "Int Mic"},
+	{"DMIC R1", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOL"},
+	{"Ext Spk", NULL, "SPOR"},
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx" },
+	{"codec_in1", NULL, "ssp2 Rx" },
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+	{"Headphone", NULL, "Platform Clock"},
+	{"Headset Mic", NULL, "Platform Clock"},
+	{"Int Mic", NULL, "Platform Clock"},
+	{"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+				  CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
+				params_rate(params) * 512, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
+
+	/* Select clk_i2s1_asrc as ASRC clock source */
+	rt5645_sel_asrc_clk_src(codec,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_DA_MONO_L_FILTER |
+				RT5645_DA_MONO_R_FILTER |
+				RT5645_AD_STEREO_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+
+	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+	if (ret < 0) {
+		dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &ctx->hp_jack,
+				    NULL, 0);
+	if (ret) {
+		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(runtime->card, "Mic Jack",
+				    SND_JACK_MICROPHONE, &ctx->mic_jack,
+				    NULL, 0);
+	if (ret) {
+		dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);
+
+	return ret;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+	.startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+	.hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5645-aif1",
+		.codec_name = "i2c-10EC5645:00",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+					| SND_SOC_DAIFMT_CBS_CFS,
+		.init = cht_codec_init,
+		.be_hw_params_fixup = cht_codec_fixup,
+		.ignore_suspend = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+	.name = "chtrt5645",
+	.dai_link = cht_dailink,
+	.num_links = ARRAY_SIZE(cht_dailink),
+	.dapm_widgets = cht_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+	.dapm_routes = cht_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	struct cht_mc_private *drv;
+
+	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
+	if (!drv)
+		return -ENOMEM;
+
+	snd_soc_card_cht.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_cht);
+	return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+	.driver = {
+		.name = "cht-bsw-rt5645",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
+MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5645");
diff --git a/kernel/sound/soc/intel/boards/cht_bsw_rt5672.c b/kernel/sound/soc/intel/boards/cht_bsw_rt5672.c
new file mode 100644
index 000000000..2c9cc5be4
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -0,0 +1,366 @@
+/*
+ *  cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
+ *                     Cherrytrail and Braswell, with RT5672 codec.
+ *
+ *  Copyright (C) 2014 Intel Corp
+ *  Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
+ *          Mengdong Lin <mengdong.lin@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.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5670.h"
+#include "../atom/sst-atom-controls.h"
+
+/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
+#define CHT_PLAT_CLK_3_HZ	19200000
+#define CHT_CODEC_DAI	"rt5670-aif1"
+
+static struct snd_soc_jack cht_bsw_headset;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin cht_bsw_headset_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headphone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+	int i;
+
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_pcm_runtime *rtd;
+
+		rtd = card->rtd + i;
+		if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+			     strlen(CHT_CODEC_DAI)))
+			return rtd->codec_dai;
+	}
+	return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret;
+
+	codec_dai = cht_get_codec_dai(card);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+		return -EIO;
+	}
+
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+		ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
+				CHT_PLAT_CLK_3_HZ, 48000 * 512);
+		if (ret < 0) {
+			dev_err(card->dev, "can't set codec pll: %d\n", ret);
+			return ret;
+		}
+
+		/* set codec sysclk source to PLL */
+		ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
+			48000 * 512, SND_SOC_CLOCK_IN);
+		if (ret < 0) {
+			dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+			return ret;
+		}
+	} else {
+		/* Set codec sysclk source to its internal clock because codec
+		 * PLL will be off when idle and MCLK will also be off by ACPI
+		 * when codec is runtime suspended. Codec needs clock for jack
+		 * detection and button press.
+		 */
+		snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
+				       48000 * 512, SND_SOC_CLOCK_IN);
+	}
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"DMIC L1", NULL, "Int Mic"},
+	{"DMIC R1", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOLP"},
+	{"Ext Spk", NULL, "SPOLN"},
+	{"Ext Spk", NULL, "SPORP"},
+	{"Ext Spk", NULL, "SPORN"},
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+	{"Headphone", NULL, "Platform Clock"},
+	{"Headset Mic", NULL, "Platform Clock"},
+	{"Int Mic", NULL, "Platform Clock"},
+	{"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
+				  CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	/* set codec sysclk source to PLL */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
+				     params_rate(params) * 512,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+	return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+	if (ret < 0) {
+		dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+		return ret;
+	}
+
+	/* Select codec ASRC clock source to track I2S1 clock, because codec
+	 * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
+	 * be supported by RT5672. Otherwise, ASRC will be disabled and cause
+	 * noise.
+	 */
+	rt5670_sel_asrc_clk_src(codec,
+				RT5670_DA_STEREO_FILTER
+				| RT5670_DA_MONO_L_FILTER
+				| RT5670_DA_MONO_R_FILTER
+				| RT5670_AD_STEREO_FILTER
+				| RT5670_AD_MONO_L_FILTER
+				| RT5670_AD_MONO_R_FILTER,
+				RT5670_CLK_SEL_I2S1_ASRC);
+
+        ret = snd_soc_card_jack_new(runtime->card, "Headset",
+                SND_JACK_HEADSET | SND_JACK_BTN_0 |
+                SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset,
+                cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins));
+        if (ret)
+                return ret;
+
+	rt5670_set_jack_detect(codec, &cht_bsw_headset);
+	return 0;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+	.startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+	.hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+	/* Front End DAI links */
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP2 - Codec */
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.nonatomic = true,
+		.codec_dai_name = "rt5670-aif1",
+		.codec_name = "i2c-10EC5670:00",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+					| SND_SOC_DAIFMT_CBS_CFS,
+		.init = cht_codec_init,
+		.be_hw_params_fixup = cht_codec_fixup,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_be_ssp2_ops,
+	},
+};
+
+static int cht_suspend_pre(struct snd_soc_card *card)
+{
+	struct snd_soc_codec *codec;
+
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+		if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
+			dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
+			rt5670_jack_suspend(codec);
+			break;
+		}
+	}
+	return 0;
+}
+
+static int cht_resume_post(struct snd_soc_card *card)
+{
+	struct snd_soc_codec *codec;
+
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+		if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
+			dev_dbg(codec->dev, "enabling jack detect for resume.\n");
+			rt5670_jack_resume(codec);
+			break;
+		}
+	}
+
+	return 0;
+}
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+	.name = "cherrytrailcraudio",
+	.dai_link = cht_dailink,
+	.num_links = ARRAY_SIZE(cht_dailink),
+	.dapm_widgets = cht_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+	.dapm_routes = cht_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+	.suspend_pre = cht_suspend_pre,
+	.resume_post = cht_resume_post,
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+
+	/* register the soc card */
+	snd_soc_card_cht.dev = &pdev->dev;
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_cht);
+	return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+	.driver = {
+		.name = "cht-bsw-rt5672",
+	},
+	.probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5672");
diff --git a/kernel/sound/soc/intel/boards/haswell.c b/kernel/sound/soc/intel/boards/haswell.c
new file mode 100644
index 000000000..22558572c
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/haswell.c
@@ -0,0 +1,209 @@
+/*
+ * Intel Haswell Lynxpoint 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 "../common/sst-dsp.h"
+#include "../haswell/sst-haswell-ipc.h"
+
+#include "../../codecs/rt5640.h"
+
+/* Haswell ULT platforms have a Headphone and Mic jack */
+static const struct snd_soc_dapm_widget haswell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route haswell_rt5640_map[] = {
+
+	{"Headphones", NULL, "HPOR"},
+	{"Headphones", NULL, "HPOL"},
+	{"IN2P", NULL, "Mic"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int haswell_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 */
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int haswell_rt5640_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, RT5640_SCLK_S_MCLK, 12288000,
+		SND_SOC_CLOCK_IN);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	/* set correct codec filter for DAI format and clock config */
+	snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000);
+
+	return ret;
+}
+
+static struct snd_soc_ops haswell_rt5640_ops = {
+	.hw_params = haswell_rt5640_hw_params,
+};
+
+static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+	struct sst_hsw *haswell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings */
+	ret = sst_hsw_device_set_config(haswell, 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, "failed to set device config\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_dai_link haswell_rt5640_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System",
+		.stream_name = "System Playback/Capture",
+		.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 = haswell_rtd_init,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.dpcm_capture = 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",
+		.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,
+	},
+
+	/* 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-INT33CA:00",
+		.codec_dai_name = "rt5640-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 = haswell_ssp0_fixup,
+		.ops = &haswell_rt5640_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */
+static struct snd_soc_card haswell_rt5640 = {
+	.name = "haswell-rt5640",
+	.owner = THIS_MODULE,
+	.dai_link = haswell_rt5640_dais,
+	.num_links = ARRAY_SIZE(haswell_rt5640_dais),
+	.dapm_widgets = haswell_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(haswell_widgets),
+	.dapm_routes = haswell_rt5640_map,
+	.num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map),
+	.fully_routed = true,
+};
+
+static int haswell_audio_probe(struct platform_device *pdev)
+{
+	haswell_rt5640.dev = &pdev->dev;
+
+	return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640);
+}
+
+static struct platform_driver haswell_audio = {
+	.probe = haswell_audio_probe,
+	.driver = {
+		.name = "haswell-audio",
+	},
+};
+
+module_platform_driver(haswell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:haswell-audio");
diff --git a/kernel/sound/soc/intel/boards/mfld_machine.c b/kernel/sound/soc/intel/boards/mfld_machine.c
new file mode 100644
index 000000000..49c09a0ad
--- /dev/null
+++ b/kernel/sound/soc/intel/boards/mfld_machine.c
@@ -0,0 +1,430 @@
+/*
+ *  mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@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.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/sn95031.h"
+
+#define MID_MONO 1
+#define MID_STEREO 2
+#define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+	MFLD_MV_START = 0,
+	/* mic bias volutage range for Headphones*/
+	MFLD_MV_HP = 400,
+	/* mic bias volutage range for American Headset*/
+	MFLD_MV_AM_HS = 650,
+	/* mic bias volutage range for Headset*/
+	MFLD_MV_HS = 2000,
+	MFLD_MV_UNDEFINED,
+};
+
+static unsigned int	hs_switch;
+static unsigned int	lo_dac;
+static struct snd_soc_codec *mfld_codec;
+
+struct mfld_mc_private {
+	void __iomem *int_base;
+	u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin = "AMIC1",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+	{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+	{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
+/* sound card controls */
+static const char *headset_switch_text[] = {"Earpiece", "Headset"};
+
+static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"};
+
+static const struct soc_enum headset_enum =
+	SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
+
+static const struct soc_enum lo_enum =
+	SOC_ENUM_SINGLE_EXT(4, lo_text);
+
+static int headset_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = hs_switch;
+	return 0;
+}
+
+static int headset_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+
+	if (ucontrol->value.integer.value[0] == hs_switch)
+		return 0;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	if (ucontrol->value.integer.value[0]) {
+		pr_debug("hs_set HS path\n");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+	} else {
+		pr_debug("hs_set EP path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
+	}
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	hs_switch = ucontrol->value.integer.value[0];
+
+	return 0;
+}
+
+static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm)
+{
+	snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
+	if (hs_switch) {
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+	} else {
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
+	}
+}
+
+static int lo_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = lo_dac;
+	return 0;
+}
+
+static int lo_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+
+	if (ucontrol->value.integer.value[0] == lo_dac)
+		return 0;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	/* we dont want to work with last state of lineout so just enable all
+	 * pins and then disable pins not required
+	 */
+	lo_enable_out_pins(dapm);
+
+	switch (ucontrol->value.integer.value[0]) {
+	case 0:
+		pr_debug("set vibra path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0);
+		break;
+
+	case 1:
+		pr_debug("set hs  path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22);
+		break;
+
+	case 2:
+		pr_debug("set spkr path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44);
+		break;
+
+	case 3:
+		pr_debug("set null path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66);
+		break;
+	}
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	lo_dac = ucontrol->value.integer.value[0];
+	return 0;
+}
+
+static const struct snd_kcontrol_new mfld_snd_controls[] = {
+	SOC_ENUM_EXT("Playback Switch", headset_enum,
+			headset_get_switch, headset_set_switch),
+	SOC_ENUM_EXT("Lineout Mux", lo_enum,
+			lo_get_switch, lo_set_switch),
+};
+
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+	{"Headphones", NULL, "HPOUTR"},
+	{"Headphones", NULL, "HPOUTL"},
+	{"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+	struct mfld_jack_data jack_data;
+
+	if (!mfld_codec)
+		return;
+
+	jack_data.mfld_jack = &mfld_jack;
+	jack_data.intr_id = intr_status;
+
+	sn95031_jack_detection(mfld_codec, &jack_data);
+	/* TODO: add american headset detection post gpiolib support */
+}
+
+static int mfld_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
+	int ret_val;
+
+	/* default is earpiece pin, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "Headphones");
+	/* default is lineout NC, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
+	lo_dac = 3;
+	hs_switch = 0;
+	/* we dont use linein in this so set to NC */
+	snd_soc_dapm_disable_pin(dapm, "LINEINL");
+	snd_soc_dapm_disable_pin(dapm, "LINEINR");
+
+	/* Headset and button jack detection */
+	ret_val = snd_soc_card_jack_new(runtime->card,
+			"Intel(R) MID Audio Jack", SND_JACK_HEADSET |
+			SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack,
+			mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins));
+	if (ret_val) {
+		pr_err("jack creation failed\n");
+		return ret_val;
+	}
+
+	ret_val = snd_soc_jack_add_zones(&mfld_jack,
+			ARRAY_SIZE(mfld_zones), mfld_zones);
+	if (ret_val) {
+		pr_err("adding jack zones failed\n");
+		return ret_val;
+	}
+
+	mfld_codec = runtime->codec;
+
+	/* we want to check if anything is inserted at boot,
+	 * so send a fake event to codec and it will read adc
+	 * to find if anything is there or not */
+	mfld_jack_check(MFLD_JACK_INSERT);
+	return ret_val;
+}
+
+static struct snd_soc_dai_link mfld_msic_dailink[] = {
+	{
+		.name = "Medfield Headset",
+		.stream_name = "Headset",
+		.cpu_dai_name = "Headset-cpu-dai",
+		.codec_dai_name = "SN95031 Headset",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = mfld_init,
+	},
+	{
+		.name = "Medfield Speaker",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Speaker-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Vibra",
+		.stream_name = "Vibra1",
+		.cpu_dai_name = "Vibra1-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra1",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Haptics",
+		.stream_name = "Vibra2",
+		.cpu_dai_name = "Vibra2-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra2",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Compress",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Compress-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_mfld = {
+	.name = "medfield_audio",
+	.owner = THIS_MODULE,
+	.dai_link = mfld_msic_dailink,
+	.num_links = ARRAY_SIZE(mfld_msic_dailink),
+
+	.controls = mfld_snd_controls,
+	.num_controls = ARRAY_SIZE(mfld_snd_controls),
+	.dapm_widgets = mfld_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mfld_widgets),
+	.dapm_routes = mfld_map,
+	.num_dapm_routes = ARRAY_SIZE(mfld_map),
+};
+
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+	struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+	memcpy_fromio(&mc_private->interrupt_status,
+			((void *)(mc_private->int_base)),
+			sizeof(u8));
+	return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+	mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+	return IRQ_HANDLED;
+}
+
+static int snd_mfld_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0, irq;
+	struct mfld_mc_private *mc_drv_ctx;
+	struct resource *irq_mem;
+
+	pr_debug("snd_mfld_mc_probe called\n");
+
+	/* retrive the irq number */
+	irq = platform_get_irq(pdev, 0);
+
+	/* audio interrupt base of SRAM location where
+	 * interrupts are stored by System FW */
+	mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
+	if (!mc_drv_ctx) {
+		pr_err("allocation failed\n");
+		return -ENOMEM;
+	}
+
+	irq_mem = platform_get_resource_byname(
+				pdev, IORESOURCE_MEM, "IRQ_BASE");
+	if (!irq_mem) {
+		pr_err("no mem resource given\n");
+		return -ENODEV;
+	}
+	mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
+						    resource_size(irq_mem));
+	if (!mc_drv_ctx->int_base) {
+		pr_err("Mapping of cache failed\n");
+		return -ENOMEM;
+	}
+	/* register for interrupt */
+	ret_val = devm_request_threaded_irq(&pdev->dev, irq,
+			snd_mfld_jack_intr_handler,
+			snd_mfld_jack_detection,
+			IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+	if (ret_val) {
+		pr_err("cannot register IRQ\n");
+		return ret_val;
+	}
+	/* register the soc card */
+	snd_soc_card_mfld.dev = &pdev->dev;
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
+	if (ret_val) {
+		pr_debug("snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, mc_drv_ctx);
+	pr_debug("successfully exited probe\n");
+	return 0;
+}
+
+static struct platform_driver snd_mfld_mc_driver = {
+	.driver = {
+		.name = "msic_audio",
+	},
+	.probe = snd_mfld_mc_probe,
+};
+
+module_platform_driver(snd_mfld_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:msic-audio");