ASoC: Rename mid-x86 directory to intel

We have other Intel platforms coming having the Smart Sound Technology (SST)
so rename the mid-x86 directory to intel as originally directory name
reflected only Intel MID platform.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

6 files changed, 1471 insertions, 0 deletions

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
new file mode 100644
index 000000000000..61c10bf503d2
--- /dev/null
+++ b/sound/soc/intel/Kconfig
@@ -0,0 +1,13 @@
+config SND_MFLD_MACHINE
+	tristate "SOC Machine Audio driver for Intel Medfield MID platform"
+	depends on INTEL_SCU_IPC
+	select SND_SOC_SN95031
+	select SND_SST_PLATFORM
+	help
+          This adds support for ASoC machine driver for Intel(R) MID Medfield platform
+          used as alsa device in audio substem in Intel(R) MID devices
+          Say Y if you have such a device
+          If unsure select "N".
+
+config SND_SST_PLATFORM
+	tristate
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
new file mode 100644
index 000000000000..639883339465
--- /dev/null
+++ b/sound/soc/intel/Makefile
@@ -0,0 +1,5 @@
+snd-soc-sst-platform-objs := sst_platform.o
+snd-soc-mfld-machine-objs := mfld_machine.o
+
+obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o
+obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o
diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c
new file mode 100644
index 000000000000..d3d4c32434f7
--- /dev/null
+++ b/sound/soc/intel/mfld_machine.c
@@ -0,0 +1,427 @@
+/*
+ *  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;
+
+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_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] == hs_switch)
+		return 0;
+
+	if (ucontrol->value.integer.value[0]) {
+		pr_debug("hs_set HS path\n");
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+	} else {
+		pr_debug("hs_set EP path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+	hs_switch = ucontrol->value.integer.value[0];
+
+	return 0;
+}
+
+static void lo_enable_out_pins(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL");
+	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR");
+	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL");
+	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR");
+	snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
+	snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
+	if (hs_switch) {
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+	} else {
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_enable_pin(&codec->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_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] == lo_dac)
+		return 0;
+
+	/* 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(codec);
+	switch (ucontrol->value.integer.value[0]) {
+	case 0:
+		pr_debug("set vibra path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT");
+		snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0);
+		break;
+
+	case 1:
+		pr_debug("set hs  path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
+		break;
+
+	case 2:
+		pr_debug("set spkr path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL");
+		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44);
+		break;
+
+	case 3:
+		pr_debug("set null path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66);
+		break;
+	}
+	snd_soc_dapm_sync(&codec->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;
+
+	jack_data.mfld_jack = &mfld_jack;
+	jack_data.intr_id = intr_status;
+
+	sn95031_jack_detection(&jack_data);
+	/* TODO: add american headset detection post gpiolib support */
+}
+
+static int mfld_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret_val;
+
+	/* Add jack sense widgets */
+	snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
+
+	/* Set up the map */
+	snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Mic");
+
+	ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls,
+				ARRAY_SIZE(mfld_snd_controls));
+	if (ret_val) {
+		pr_err("soc_add_controls failed %d", ret_val);
+		return 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_jack_new(codec, "Intel(R) MID Audio Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 |
+			SND_JACK_BTN_1, &mfld_jack);
+	if (ret_val) {
+		pr_err("jack creation failed\n");
+		return ret_val;
+	}
+
+	ret_val = snd_soc_jack_add_pins(&mfld_jack,
+			ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
+	if (ret_val) {
+		pr_err("adding jack pins 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;
+	}
+
+	/* 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),
+};
+
+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;
+
+	if (mfld_jack.codec == NULL)
+		return IRQ_HANDLED;
+	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 = {
+		.owner = THIS_MODULE,
+		.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");
diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst_dsp.h
new file mode 100644
index 000000000000..0fce1de284ff
--- /dev/null
+++ b/sound/soc/intel/sst_dsp.h
@@ -0,0 +1,134 @@
+#ifndef __SST_DSP_H__
+#define __SST_DSP_H__
+/*
+ *  sst_dsp.h - Intel SST Driver for audio engine
+ *
+ *  Copyright (C) 2008-12 Intel Corporation
+ *  Authors:	Vinod Koul <vinod.koul@linux.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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+enum sst_codec_types {
+	/*  AUDIO/MUSIC	CODEC Type Definitions */
+	SST_CODEC_TYPE_UNKNOWN = 0,
+	SST_CODEC_TYPE_PCM,	/* Pass through Audio codec */
+	SST_CODEC_TYPE_MP3,
+	SST_CODEC_TYPE_MP24,
+	SST_CODEC_TYPE_AAC,
+	SST_CODEC_TYPE_AACP,
+	SST_CODEC_TYPE_eAACP,
+};
+
+enum stream_type {
+	SST_STREAM_TYPE_NONE = 0,
+	SST_STREAM_TYPE_MUSIC = 1,
+};
+
+struct snd_pcm_params {
+	u16 codec;	/* codec type */
+	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 reserved2;
+	u16 reserved3;
+	u8 channel_map[8];
+} __packed;
+
+/* MP3 Music Parameters Message */
+struct snd_mp3_params {
+	u16 codec;
+	u8  num_chan;	/* 1=Mono, 2=Stereo	*/
+	u8  pcm_wd_sz; /* 16/24 - bit*/
+	u8  crc_check; /* crc_check - disable (0) or enable (1) */
+	u8  reserved1; /* unused*/
+	u16 reserved2;	/* Unused */
+} __packed;
+
+#define AAC_BIT_STREAM_ADTS		0
+#define AAC_BIT_STREAM_ADIF		1
+#define AAC_BIT_STREAM_RAW		2
+
+/* AAC Music Parameters Message */
+struct snd_aac_params {
+	u16 codec;
+	u8 num_chan; /* 1=Mono, 2=Stereo*/
+	u8 pcm_wd_sz; /* 16/24 - bit*/
+	u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
+	u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
+	u16  reser2;
+	u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
+	u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
+	u8 reser1;
+	u16  reser3;
+} __packed;
+
+/* WMA Music Parameters Message */
+struct snd_wma_params {
+	u16 codec;
+	u8  num_chan;	/* 1=Mono, 2=Stereo */
+	u8  pcm_wd_sz;	/* 16/24 - bit*/
+	u32 brate;	/* Use the hard coded value. */
+	u32 sfreq;	/* Sampling freq eg. 8000, 441000, 48000 */
+	u32 channel_mask;  /* Channel Mask */
+	u16 format_tag;	/* Format Tag */
+	u16 block_align;	/* packet size */
+	u16 wma_encode_opt;/* Encoder option */
+	u8 op_align;	/* op align 0- 16 bit, 1- MSB, 2 LSB */
+	u8 reserved;	/* reserved */
+} __packed;
+
+/* Codec params struture */
+union  snd_sst_codec_params {
+	struct snd_pcm_params pcm_params;
+	struct snd_mp3_params mp3_params;
+	struct snd_aac_params aac_params;
+	struct snd_wma_params wma_params;
+} __packed;
+
+/* Address and size info of a frame buffer */
+struct sst_address_info {
+	u32 addr; /* Address at IA */
+	u32 size; /* Size of the buffer */
+};
+
+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
+				  message is sent valid only if path  = 0*/
+} __packed;
+
+struct snd_sst_stream_params {
+	union snd_sst_codec_params uc;
+} __packed;
+
+struct snd_sst_params {
+	u32 stream_id;
+	u8 codec;
+	u8 ops;
+	u8 stream_type;
+	u8 device_type;
+	struct snd_sst_stream_params sparams;
+	struct snd_sst_alloc_params_ext aparams;
+};
+
+#endif /* __SST_DSP_H__ */
diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c
new file mode 100644
index 000000000000..b6b5eb698d33
--- /dev/null
+++ b/sound/soc/intel/sst_platform.c
@@ -0,0 +1,735 @@
+/*
+ *  sst_platform.c - Intel MID Platform driver
+ *
+ *  Copyright (C) 2010-2013 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/slab.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/compress_driver.h>
+#include "sst_platform.h"
+
+static struct sst_device *sst;
+static DEFINE_MUTEX(sst_lock);
+
+int sst_register_dsp(struct sst_device *dev)
+{
+	if (WARN_ON(!dev))
+		return -EINVAL;
+	if (!try_module_get(dev->dev->driver->owner))
+		return -ENODEV;
+	mutex_lock(&sst_lock);
+	if (sst) {
+		pr_err("we already have a device %s\n", sst->name);
+		module_put(dev->dev->driver->owner);
+		mutex_unlock(&sst_lock);
+		return -EEXIST;
+	}
+	pr_debug("registering device %s\n", dev->name);
+	sst = dev;
+	mutex_unlock(&sst_lock);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(sst_register_dsp);
+
+int sst_unregister_dsp(struct sst_device *dev)
+{
+	if (WARN_ON(!dev))
+		return -EINVAL;
+	if (dev != sst)
+		return -EINVAL;
+
+	mutex_lock(&sst_lock);
+
+	if (!sst) {
+		mutex_unlock(&sst_lock);
+		return -EIO;
+	}
+
+	module_put(sst->dev->driver->owner);
+	pr_debug("unreg %s\n", sst->name);
+	sst = NULL;
+	mutex_unlock(&sst_lock);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(sst_unregister_dsp);
+
+static struct snd_pcm_hardware sst_platform_pcm_hw = {
+	.info =	(SNDRV_PCM_INFO_INTERLEAVED |
+			SNDRV_PCM_INFO_DOUBLE |
+			SNDRV_PCM_INFO_PAUSE |
+			SNDRV_PCM_INFO_RESUME |
+			SNDRV_PCM_INFO_MMAP|
+			SNDRV_PCM_INFO_MMAP_VALID |
+			SNDRV_PCM_INFO_BLOCK_TRANSFER |
+			SNDRV_PCM_INFO_SYNC_START),
+	.formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 |
+			SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 |
+			SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32),
+	.rates = (SNDRV_PCM_RATE_8000|
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000),
+	.rate_min = SST_MIN_RATE,
+	.rate_max = SST_MAX_RATE,
+	.channels_min =	SST_MIN_CHANNEL,
+	.channels_max =	SST_MAX_CHANNEL,
+	.buffer_bytes_max = SST_MAX_BUFFER,
+	.period_bytes_min = SST_MIN_PERIOD_BYTES,
+	.period_bytes_max = SST_MAX_PERIOD_BYTES,
+	.periods_min = SST_MIN_PERIODS,
+	.periods_max = SST_MAX_PERIODS,
+	.fifo_size = SST_FIFO_SIZE,
+};
+
+/* MFLD - MSIC */
+static struct snd_soc_dai_driver sst_platform_dai[] = {
+{
+	.name = "Headset-cpu-dai",
+	.id = 0,
+	.playback = {
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 5,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Speaker-cpu-dai",
+	.id = 1,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Vibra1-cpu-dai",
+	.id = 2,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_MONO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Vibra2-cpu-dai",
+	.id = 3,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Compress-cpu-dai",
+	.compress_dai = 1,
+	.playback = {
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+};
+
+static const struct snd_soc_component_driver sst_component = {
+	.name		= "sst",
+};
+
+/* helper functions */
+static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
+					int state)
+{
+	unsigned long flags;
+	spin_lock_irqsave(&stream->status_lock, flags);
+	stream->stream_status = state;
+	spin_unlock_irqrestore(&stream->status_lock, flags);
+}
+
+static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
+{
+	int state;
+	unsigned long flags;
+
+	spin_lock_irqsave(&stream->status_lock, flags);
+	state = stream->stream_status;
+	spin_unlock_irqrestore(&stream->status_lock, flags);
+	return state;
+}
+
+static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
+				struct sst_pcm_params *param)
+{
+
+	param->codec = SST_CODEC_TYPE_PCM;
+	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)
+{
+	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;
+
+	/* set codec params and inform SST driver the same */
+	sst_fill_pcm_params(substream, &param);
+	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);
+	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);
+	return ret_val;
+}
+
+static void sst_period_elapsed(void *mad_substream)
+{
+	struct snd_pcm_substream *substream = mad_substream;
+	struct sst_runtime_stream *stream;
+	int status;
+
+	if (!substream || !substream->runtime)
+		return;
+	stream = substream->runtime->private_data;
+	if (!stream)
+		return;
+	status = sst_get_stream_status(stream);
+	if (status != SST_PLATFORM_RUNNING)
+		return;
+	snd_pcm_period_elapsed(substream);
+}
+
+static int sst_platform_init_stream(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	int ret_val;
+
+	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.buffer_ptr = 0;
+	stream->stream_info.sfreq = substream->runtime->rate;
+	ret_val = stream->ops->device_control(
+			SST_SND_STREAM_INIT, &stream->stream_info);
+	if (ret_val)
+		pr_err("control_set ret error %d\n", ret_val);
+	return ret_val;
+
+}
+/* end -- helper functions */
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+	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)
+		return -ENOMEM;
+	spin_lock_init(&stream->status_lock);
+
+	/* get the sst ops */
+	mutex_lock(&sst_lock);
+	if (!sst) {
+		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;
+	}
+	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;
+	/* allocate memory for SST API set */
+	runtime->private_data = stream;
+
+	return 0;
+}
+
+static int sst_platform_close(struct snd_pcm_substream *substream)
+{
+	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)
+{
+	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) {
+		ret_val = stream->ops->device_control(
+				SST_SND_DROP, &str_id);
+		return ret_val;
+	}
+
+	ret_val = sst_platform_alloc_stream(substream);
+	if (ret_val < 0)
+		return ret_val;
+	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
+			"%d", stream->stream_info.str_id);
+
+	ret_val = sst_platform_init_stream(substream);
+	if (ret_val)
+		return ret_val;
+	substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER;
+	return ret_val;
+}
+
+static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
+					int cmd)
+{
+	int ret_val = 0, str_id;
+	struct sst_runtime_stream *stream;
+	int str_cmd, status;
+
+	pr_debug("sst_platform_pcm_trigger called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		pr_debug("sst: Trigger Start\n");
+		str_cmd = SST_SND_START;
+		status = SST_PLATFORM_RUNNING;
+		stream->stream_info.mad_substream = substream;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		pr_debug("sst: in stop\n");
+		str_cmd = SST_SND_DROP;
+		status = SST_PLATFORM_DROPPED;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		pr_debug("sst: in pause\n");
+		str_cmd = SST_SND_PAUSE;
+		status = SST_PLATFORM_PAUSED;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		pr_debug("sst: in pause release\n");
+		str_cmd = SST_SND_RESUME;
+		status = SST_PLATFORM_RUNNING;
+		break;
+	default:
+		return -EINVAL;
+	}
+	ret_val = stream->ops->device_control(str_cmd, &str_id);
+	if (!ret_val)
+		sst_set_stream_status(stream, status);
+
+	return ret_val;
+}
+
+
+static snd_pcm_uframes_t sst_platform_pcm_pointer
+			(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val, status;
+	struct pcm_stream_info *str_info;
+
+	stream = substream->runtime->private_data;
+	status = sst_get_stream_status(stream);
+	if (status == SST_PLATFORM_INIT)
+		return 0;
+	str_info = &stream->stream_info;
+	ret_val = stream->ops->device_control(
+				SST_SND_BUFFER_POINTER, str_info);
+	if (ret_val) {
+		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);
+}
+
+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)
+{
+	pr_debug("sst_pcm_free called\n");
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_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) {
+		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
+			SNDRV_DMA_TYPE_CONTINUOUS,
+			snd_dma_continuous_data(GFP_KERNEL),
+			SST_MIN_BUFFER, SST_MAX_BUFFER);
+		if (retval) {
+			pr_err("dma buffer allocationf fail\n");
+			return retval;
+		}
+	}
+	return retval;
+}
+
+/* compress stream operations */
+static void sst_compr_fragment_elapsed(void *arg)
+{
+	struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg;
+
+	pr_debug("fragment elapsed by driver\n");
+	if (cstream)
+		snd_compr_fragment_elapsed(cstream);
+}
+
+static int sst_platform_compr_open(struct snd_compr_stream *cstream)
+{
+
+	int ret_val = 0;
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct sst_runtime_stream *stream;
+
+	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+	if (!stream)
+		return -ENOMEM;
+
+	spin_lock_init(&stream->status_lock);
+
+	/* get the sst ops */
+	if (!sst || !try_module_get(sst->dev->driver->owner)) {
+		pr_err("no device available to run\n");
+		ret_val = -ENODEV;
+		goto out_ops;
+	}
+	stream->compr_ops = sst->compr_ops;
+
+	stream->id = 0;
+	sst_set_stream_status(stream, SST_PLATFORM_INIT);
+	runtime->private_data = stream;
+	return 0;
+out_ops:
+	kfree(stream);
+	return ret_val;
+}
+
+static int sst_platform_compr_free(struct snd_compr_stream *cstream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val = 0, str_id;
+
+	stream = cstream->runtime->private_data;
+	/*need to check*/
+	str_id = stream->id;
+	if (str_id)
+		ret_val = stream->compr_ops->close(str_id);
+	module_put(sst->dev->driver->owner);
+	kfree(stream);
+	pr_debug("%s: %d\n", __func__, ret_val);
+	return 0;
+}
+
+static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
+					struct snd_compr_params *params)
+{
+	struct sst_runtime_stream *stream;
+	int retval;
+	struct snd_sst_params str_params;
+	struct sst_compress_cb cb;
+
+	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;
+
+	switch (params->codec.id) {
+	case SND_AUDIOCODEC_MP3: {
+		str_params.codec = SST_CODEC_TYPE_MP3;
+		str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3;
+		str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in;
+		str_params.sparams.uc.mp3_params.pcm_wd_sz = 16;
+		break;
+	}
+
+	case SND_AUDIOCODEC_AAC: {
+		str_params.codec = SST_CODEC_TYPE_AAC;
+		str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC;
+		str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in;
+		str_params.sparams.uc.aac_params.pcm_wd_sz = 16;
+		if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS)
+			str_params.sparams.uc.aac_params.bs_format =
+							AAC_BIT_STREAM_ADTS;
+		else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW)
+			str_params.sparams.uc.aac_params.bs_format =
+							AAC_BIT_STREAM_RAW;
+		else {
+			pr_err("Undefined format%d\n", params->codec.format);
+			return -EINVAL;
+		}
+		str_params.sparams.uc.aac_params.externalsr =
+						params->codec.sample_rate;
+		break;
+	}
+
+	default:
+		pr_err("codec not supported, id =%d\n", params->codec.id);
+		return -EINVAL;
+	}
+
+	str_params.aparams.ring_buf_info[0].addr  =
+					virt_to_phys(cstream->runtime->buffer);
+	str_params.aparams.ring_buf_info[0].size =
+					cstream->runtime->buffer_size;
+	str_params.aparams.sg_count = 1;
+	str_params.aparams.frag_size = cstream->runtime->fragment_size;
+
+	cb.param = cstream;
+	cb.compr_cb = sst_compr_fragment_elapsed;
+
+	retval = stream->compr_ops->open(&str_params, &cb);
+	if (retval < 0) {
+		pr_err("stream allocation failed %d\n", retval);
+		return retval;
+	}
+
+	stream->id = retval;
+	return 0;
+}
+
+static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+	struct sst_runtime_stream *stream =
+		cstream->runtime->private_data;
+
+	return stream->compr_ops->control(cmd, stream->id);
+}
+
+static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
+					struct snd_compr_tstamp *tstamp)
+{
+	struct sst_runtime_stream *stream;
+
+	stream  = cstream->runtime->private_data;
+	stream->compr_ops->tstamp(stream->id, tstamp);
+	tstamp->byte_offset = tstamp->copied_total %
+				 (u32)cstream->runtime->buffer_size;
+	pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
+	return 0;
+}
+
+static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
+					size_t bytes)
+{
+	struct sst_runtime_stream *stream;
+
+	stream  = cstream->runtime->private_data;
+	stream->compr_ops->ack(stream->id, (unsigned long)bytes);
+	stream->bytes_written += bytes;
+
+	return 0;
+}
+
+static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream,
+					struct snd_compr_caps *caps)
+{
+	struct sst_runtime_stream *stream =
+		cstream->runtime->private_data;
+
+	return stream->compr_ops->get_caps(caps);
+}
+
+static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
+					struct snd_compr_codec_caps *codec)
+{
+	struct sst_runtime_stream *stream =
+		cstream->runtime->private_data;
+
+	return stream->compr_ops->get_codec_caps(codec);
+}
+
+static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream,
+					struct snd_compr_metadata *metadata)
+{
+	struct sst_runtime_stream *stream  =
+		 cstream->runtime->private_data;
+
+	return stream->compr_ops->set_metadata(stream->id, metadata);
+}
+
+static struct snd_compr_ops sst_platform_compr_ops = {
+
+	.open = sst_platform_compr_open,
+	.free = sst_platform_compr_free,
+	.set_params = sst_platform_compr_set_params,
+	.set_metadata = sst_platform_compr_set_metadata,
+	.trigger = sst_platform_compr_trigger,
+	.pointer = sst_platform_compr_pointer,
+	.ack = sst_platform_compr_ack,
+	.get_caps = sst_platform_compr_get_caps,
+	.get_codec_caps = sst_platform_compr_get_codec_caps,
+};
+
+static struct snd_soc_platform_driver sst_soc_platform_drv = {
+	.ops		= &sst_platform_ops,
+	.compr_ops	= &sst_platform_compr_ops,
+	.pcm_new	= sst_pcm_new,
+	.pcm_free	= sst_pcm_free,
+};
+
+static int sst_platform_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	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");
+		return ret;
+	}
+
+	ret = snd_soc_register_component(&pdev->dev, &sst_component,
+				sst_platform_dai, ARRAY_SIZE(sst_platform_dai));
+	if (ret) {
+		pr_err("registering cpu dais failed\n");
+		snd_soc_unregister_platform(&pdev->dev);
+	}
+	return ret;
+}
+
+static int sst_platform_remove(struct platform_device *pdev)
+{
+
+	snd_soc_unregister_component(&pdev->dev);
+	snd_soc_unregister_platform(&pdev->dev);
+	pr_debug("sst_platform_remove success\n");
+	return 0;
+}
+
+static struct platform_driver sst_platform_driver = {
+	.driver		= {
+		.name		= "sst-platform",
+		.owner		= THIS_MODULE,
+	},
+	.probe		= sst_platform_probe,
+	.remove		= sst_platform_remove,
+};
+
+module_platform_driver(sst_platform_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sst-platform");
diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h
new file mode 100644
index 000000000000..cacc9066ec52
--- /dev/null
+++ b/sound/soc/intel/sst_platform.h
@@ -0,0 +1,157 @@
+/*
+ *  sst_platform.h - Intel MID Platform driver header file
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+
+#ifndef __SST_PLATFORMDRV_H__
+#define __SST_PLATFORMDRV_H__
+
+#include "sst_dsp.h"
+
+#define SST_MONO		1
+#define SST_STEREO		2
+#define SST_MAX_CAP		5
+
+#define SST_MIN_RATE		8000
+#define SST_MAX_RATE		48000
+#define SST_MIN_CHANNEL		1
+#define SST_MAX_CHANNEL		5
+#define SST_MAX_BUFFER		(800*1024)
+#define SST_MIN_BUFFER		(800*1024)
+#define SST_MIN_PERIOD_BYTES	32
+#define SST_MAX_PERIOD_BYTES	SST_MAX_BUFFER
+#define SST_MIN_PERIODS		2
+#define SST_MAX_PERIODS		(1024*2)
+#define SST_FIFO_SIZE		0
+
+struct pcm_stream_info {
+	int str_id;
+	void *mad_substream;
+	void (*period_elapsed) (void *mad_substream);
+	unsigned long long buffer_ptr;
+	int sfreq;
+};
+
+enum sst_drv_status {
+	SST_PLATFORM_INIT = 1,
+	SST_PLATFORM_STARTED,
+	SST_PLATFORM_RUNNING,
+	SST_PLATFORM_PAUSED,
+	SST_PLATFORM_DROPPED,
+};
+
+enum sst_controls {
+	SST_SND_ALLOC =			0x00,
+	SST_SND_PAUSE =			0x01,
+	SST_SND_RESUME =		0x02,
+	SST_SND_DROP =			0x03,
+	SST_SND_FREE =			0x04,
+	SST_SND_BUFFER_POINTER =	0x05,
+	SST_SND_STREAM_INIT =		0x06,
+	SST_SND_START	 =		0x07,
+	SST_MAX_CONTROLS =		0x07,
+};
+
+enum sst_stream_ops {
+	STREAM_OPS_PLAYBACK = 0,
+	STREAM_OPS_CAPTURE,
+};
+
+enum sst_audio_device_type {
+	SND_SST_DEVICE_HEADSET = 1,
+	SND_SST_DEVICE_IHF,
+	SND_SST_DEVICE_VIBRA,
+	SND_SST_DEVICE_HAPTIC,
+	SND_SST_DEVICE_CAPTURE,
+	SND_SST_DEVICE_COMPRESS,
+};
+
+/* PCM Parameters */
+struct sst_pcm_params {
+	u16 codec;	/* codec type */
+	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 */
+	u32 ring_buffer_size;
+	u32 period_count;	/* period elapsed in samples*/
+	u32 ring_buffer_addr;
+};
+
+struct sst_stream_params {
+	u32 result;
+	u32 stream_id;
+	u8 codec;
+	u8 ops;
+	u8 stream_type;
+	u8 device_type;
+	struct sst_pcm_params sparams;
+};
+
+struct sst_compress_cb {
+	void *param;
+	void (*compr_cb)(void *param);
+};
+
+struct compress_sst_ops {
+	const char *name;
+	int (*open) (struct snd_sst_params *str_params,
+			struct sst_compress_cb *cb);
+	int (*control) (unsigned int cmd, unsigned int str_id);
+	int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
+	int (*ack) (unsigned int str_id, unsigned long bytes);
+	int (*close) (unsigned int str_id);
+	int (*get_caps) (struct snd_compr_caps *caps);
+	int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
+	int (*set_metadata) (unsigned int str_id,
+			struct snd_compr_metadata *mdata);
+
+};
+
+struct sst_ops {
+	int (*open) (struct sst_stream_params *str_param);
+	int (*device_control) (int cmd, void *arg);
+	int (*close) (unsigned int str_id);
+};
+
+struct sst_runtime_stream {
+	int     stream_status;
+	unsigned int id;
+	size_t bytes_written;
+	struct pcm_stream_info stream_info;
+	struct sst_ops *ops;
+	struct compress_sst_ops *compr_ops;
+	spinlock_t	status_lock;
+};
+
+struct sst_device {
+	char *name;
+	struct device *dev;
+	struct sst_ops *ops;
+	struct compress_sst_ops *compr_ops;
+};
+
+int sst_register_dsp(struct sst_device *sst);
+int sst_unregister_dsp(struct sst_device *sst);
+#endif