Merge tag 'asoc-v3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v3.17

This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC.  A big thanks to Benoit and Misael for their work on
this.

Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.

 - Support for multiple CODECs attached to a single DAI, enabling
   systems with for example multiple DAC/speaker drivers on a single
   link, contributed by Benoit Cousson based on work from Misael Lopez
   Cruz.
 - Support for byte controls larger than 256 bytes based on the use of
   TLVs contributed by Omair Mohammed Abdullah.
 - More componentisation work from Lars-Peter Clausen.
 - The remainder of the conversions of CODEC drivers to params_width()
 - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
   RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
   TAS2552.
 - Lots of updates and fixes, especially to the DaVinci, Intel,
   Freescale, Realtek, and rcar drivers.

17 files changed, 1158 insertions, 171 deletions

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index c30fedb3e149..f5b4a9c79cdf 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
 	help
 	  This adds audio driver for Intel Baytrail platform based boards
 	  with the MAX98090 audio codec.
+
+config SND_SOC_INTEL_BROADWELL_MACH
+	tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
+	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
+	select SND_SOC_INTEL_HASWELL
+	select SND_COMPRESS_OFFLOAD
+	select SND_SOC_RT286
+	help
+	  This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
+	  Ultrabook platforms.
+	  Say Y if you have such a device
+	  If unsure select "N".
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 4bfca79a42ba..7acbfc43a0c6 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o
 snd-soc-sst-haswell-objs := haswell.o
 snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
 snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-broadwell-objs := broadwell.o
 
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c
new file mode 100644
index 000000000000..0e550f14028f
--- /dev/null
+++ b/sound/soc/intel/broadwell.c
@@ -0,0 +1,251 @@
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt286.h"
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("DMIC1", NULL),
+	SND_SOC_DAPM_MIC("DMIC2", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack deteck */
+	{"Headphones", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+	{"LINE1", NULL, "Line Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC1"},
+	{"DMIC2 Pin", NULL, "DMIC2"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The ADSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+		SND_SOC_CLOCK_IN);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+	.hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+	struct sst_hsw *broadwell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings */
+	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+		SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	if (ret < 0) {
+		dev_err(rtd->dev, "error: failed to set device config\n");
+		return ret;
+	}
+
+	/* always connected - check HP for jack detect */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line Jack");
+	snd_soc_dapm_enable_pin(dapm, "DMIC1");
+	snd_soc_dapm_enable_pin(dapm, "DMIC2");
+
+	return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System PCM",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = broadwell_rtd_init,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload0",
+		.stream_name = "Offload0 Playback",
+		.cpu_dai_name = "Offload0 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload1",
+		.stream_name = "Offload1 Playback",
+		.cpu_dai_name = "Offload1 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Loopback PCM",
+		.stream_name = "Loopback",
+		.cpu_dai_name = "Loopback Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 0,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "Capture PCM",
+		.stream_name = "Capture",
+		.cpu_dai_name = "Capture Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "Codec",
+		.be_id = 0,
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt286-aif1",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broadwell_ssp0_fixup,
+		.ops = &broadwell_rt286_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+	.name = "broadwell-rt286",
+	.owner = THIS_MODULE,
+	.dai_link = broadwell_rt286_dais,
+	.num_links = ARRAY_SIZE(broadwell_rt286_dais),
+	.dapm_widgets = broadwell_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+	.dapm_routes = broadwell_rt286_map,
+	.num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+	.fully_routed = true,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+	broadwell_rt286.dev = &pdev->dev;
+
+	return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&broadwell_rt286);
+	return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+	.probe = broadwell_audio_probe,
+	.remove = broadwell_audio_remove,
+	.driver = {
+		.name = "broadwell-audio",
+		.owner = THIS_MODULE,
+	},
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");
diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c
index 5fc98c64a3f4..b8b8af571ef1 100644
--- a/sound/soc/intel/byt-max98090.c
+++ b/sound/soc/intel/byt-max98090.c
@@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
 
 static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
 	{"IN34", NULL, "Headset Mic"},
-	{"IN34", NULL, "MICBIAS"},
-	{"MICBIAS", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "MICBIAS"},
 	{"DMICL", NULL, "Int Mic"},
 	{"Headphone", NULL, "HPL"},
 	{"Headphone", NULL, "HPR"},
@@ -64,14 +63,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = {
 		.pin	= "Headset Mic",
 		.mask	= SND_JACK_MICROPHONE,
 	},
-	{
-		.pin	= "Ext Spk",
-		.mask	= SND_JACK_LINEOUT,
-	},
-	{
-		.pin	= "Int Mic",
-		.mask	= SND_JACK_LINEIN,
-	},
 };
 
 static struct snd_soc_jack_gpio hs_jack_gpios[] = {
@@ -84,7 +75,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
 	{
 		.name		= "mic-gpio",
 		.idx		= 1,
-		.report		= SND_JACK_MICROPHONE | SND_JACK_LINEIN,
+		.invert		= 1,
+		.report		= SND_JACK_MICROPHONE,
 		.debounce_time	= 200,
 	},
 };
@@ -108,7 +100,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
 	}
 
 	/* Enable jack detection */
-	ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack);
+	ret = snd_soc_jack_new(codec, "Headset",
+			       SND_JACK_LINEOUT | SND_JACK_HEADSET, jack);
 	if (ret)
 		return ret;
 
@@ -117,13 +110,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
 	if (ret)
 		return ret;
 
-	ret = snd_soc_jack_add_gpiods(card->dev->parent, jack,
-				      ARRAY_SIZE(hs_jack_gpios),
-				      hs_jack_gpios);
-	if (ret)
-		return ret;
-
-	return max98090_mic_detect(codec, jack);
+	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[] = {
diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c
index 53d160d39972..234a58de3c53 100644
--- a/sound/soc/intel/byt-rt5640.c
+++ b/sound/soc/intel/byt-rt5640.c
@@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+	{"Headset Mic", NULL, "MICBIAS1"},
 	{"IN2P", NULL, "Headset Mic"},
 	{"IN2N", NULL, "Headset Mic"},
 	{"DMIC1", NULL, "Internal Mic"},
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
new file mode 100644
index 000000000000..14063ab8c7c5
--- /dev/null
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (C) 2013-14 Intel Corp
+ *  Author: Ramesh Babu <ramesh.babu.koul@intel.com>
+ *  	Omair M Abdullah <omair.m.abdullah@intel.com>
+ *  	Samreen Nilofer <samreen.nilofer@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+
+#ifndef __SST_CONTROLS_V2_H__
+#define __SST_CONTROLS_V2_H__
+
+enum {
+	MERR_DPCM_AUDIO = 0,
+	MERR_DPCM_COMPR,
+};
+
+
+#endif
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index d207b22ea330..67673a2c0f41 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -122,6 +122,26 @@ struct sst_byt_tstamp {
 	u32 channel_peak[8];
 } __packed;
 
+struct sst_byt_fw_version {
+	u8 build;
+	u8 minor;
+	u8 major;
+	u8 type;
+} __packed;
+
+struct sst_byt_fw_build_info {
+	u8 date[16];
+	u8 time[16];
+} __packed;
+
+struct sst_byt_fw_init {
+	struct sst_byt_fw_version fw_version;
+	struct sst_byt_fw_build_info build_info;
+	u16 result;
+	u8 module_id;
+	u8 debug_info;
+} __packed;
+
 /* driver internal IPC message structure */
 struct ipc_message {
 	struct list_head list;
@@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_byt *byt;
 	struct sst_fw *byt_sst_fw;
+	struct sst_byt_fw_init init;
 	int err;
 
 	dev_dbg(dev, "initialising Byt DSP IPC\n");
@@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 		goto boot_err;
 	}
 
+	/* show firmware information */
+	sst_dsp_inbox_read(byt->dsp, &init, sizeof(init));
+	dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n",
+		 init.fw_version.major, init.fw_version.minor,
+		 init.fw_version.build, init.fw_version.type);
+	dev_info(byt->dev, "Build type: %x\n", init.fw_version.type);
+	dev_info(byt->dev, "Build date: %s %s\n",
+		 init.build_info.date, init.build_info.time);
+
 	pdata->dsp = byt;
 	byt->fw = byt_sst_fw;
 
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 8eab97368ea7..599401c0c655 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-				  SNDRV_PCM_FORMAT_S24_LE,
+				  SNDRV_PCM_FMTBIT_S24_LE,
 	.period_bytes_min	= 384,
 	.period_bytes_max	= 48000,
 	.periods_min		= 2,
diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c
index 0b715b20a2d7..cd23060a0d86 100644
--- a/sound/soc/intel/sst-dsp.c
+++ b/sound/soc/intel/sst-dsp.c
@@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64);
 
 void sst_dsp_dump(struct sst_dsp *sst)
 {
-	sst->ops->dump(sst);
+	if (sst->ops->dump)
+		sst->ops->dump(sst);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_dump);
 
 void sst_dsp_reset(struct sst_dsp *sst)
 {
-	sst->ops->reset(sst);
+	if (sst->ops->reset)
+		sst->ops->reset(sst);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_reset);
 
 int sst_dsp_boot(struct sst_dsp *sst)
 {
-	sst->ops->boot(sst);
+	if (sst->ops->boot)
+		sst->ops->boot(sst);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(sst_dsp_boot);
diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h
index e44423be66c4..3165dfa97408 100644
--- a/sound/soc/intel/sst-dsp.h
+++ b/sound/soc/intel/sst-dsp.h
@@ -52,7 +52,11 @@
 #define SST_CLKCTL		0x78
 #define SST_CSR2		0x80
 #define SST_LTRC		0xE0
-#define SST_HDMC		0xE8
+#define SST_HMDC		0xE8
+
+#define SST_SHIM_BEGIN		SST_CSR
+#define SST_SHIM_END		SST_HDMC
+
 #define SST_DBGO		0xF0
 
 #define SST_SHIM_SIZE		0x100
@@ -73,6 +77,8 @@
 #define SST_CSR_S0IOCS		(0x1 << 21)
 #define SST_CSR_S1IOCS		(0x1 << 23)
 #define SST_CSR_LPCS		(0x1 << 31)
+#define SST_CSR_24MHZ_LPCS	(SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS)
+#define SST_CSR_24MHZ_NO_LPCS	(SST_CSR_SBCS0 | SST_CSR_SBCS1)
 #define SST_BYT_CSR_RST		(0x1 << 0)
 #define SST_BYT_CSR_VECTOR_SEL	(0x1 << 1)
 #define SST_BYT_CSR_STALL	(0x1 << 2)
@@ -92,6 +98,14 @@
 #define SST_IMRX_DONE		(0x1 << 0)
 #define SST_BYT_IMRX_REQUEST	(0x1 << 1)
 
+/* IMRD / IMD */
+#define SST_IMRD_DONE		(0x1 << 0)
+#define SST_IMRD_BUSY		(0x1 << 1)
+#define SST_IMRD_SSP0		(0x1 << 16)
+#define SST_IMRD_DMAC0		(0x1 << 21)
+#define SST_IMRD_DMAC1		(0x1 << 22)
+#define SST_IMRD_DMAC		(SST_IMRD_DMAC0 | SST_IMRD_DMAC1)
+
 /*  IPCX / IPCC */
 #define	SST_IPCX_DONE		(0x1 << 30)
 #define	SST_IPCX_BUSY		(0x1 << 31)
@@ -118,9 +132,21 @@
 /* LTRC */
 #define SST_LTRC_VAL(x)		(x << 0)
 
-/* HDMC */
-#define SST_HDMC_HDDA0(x)	(x << 0)
-#define SST_HDMC_HDDA1(x)	(x << 7)
+/* HMDC */
+#define SST_HMDC_HDDA0(x)	(x << 0)
+#define SST_HMDC_HDDA1(x)	(x << 7)
+#define SST_HMDC_HDDA_E0_CH0	1
+#define SST_HMDC_HDDA_E0_CH1	2
+#define SST_HMDC_HDDA_E0_CH2	4
+#define SST_HMDC_HDDA_E0_CH3	8
+#define SST_HMDC_HDDA_E1_CH0	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0)
+#define SST_HMDC_HDDA_E1_CH1	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1)
+#define SST_HMDC_HDDA_E1_CH2	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2)
+#define SST_HMDC_HDDA_E1_CH3	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E0_ALLCH	(SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \
+				 SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E1_ALLCH	(SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \
+				 SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3)
 
 
 /* SST Vendor Defined Registers and bits */
@@ -130,11 +156,16 @@
 #define SST_VDRTCTL3		0xaC
 
 /* VDRTCTL0 */
+#define SST_VDRTCL0_APLLSE_MASK		1
 #define SST_VDRTCL0_DSRAMPGE_SHIFT	16
 #define SST_VDRTCL0_DSRAMPGE_MASK	(0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
 #define SST_VDRTCL0_ISRAMPGE_SHIFT	6
 #define SST_VDRTCL0_ISRAMPGE_MASK	(0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
 
+/* PMCS */
+#define SST_PMCS		0x84
+#define SST_PMCS_PS_MASK	0x3
+
 struct sst_dsp;
 
 /*
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index 535f517629fd..4b6c163c10ff 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -28,9 +28,6 @@
 #include <linux/firmware.h>
 #include <linux/pm_runtime.h>
 
-#include <linux/acpi.h>
-#include <acpi/acpi_bus.h>
-
 #include "sst-dsp.h"
 #include "sst-dsp-priv.h"
 #include "sst-haswell-ipc.h"
@@ -272,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst)
 		SST_CSR2_SDFD_SSP1);
 
 	/* enable DMA engine 0,1 all channels to access host memory */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
-		SST_HDMC_HDDA1(0xff)  | SST_HDMC_HDDA0(0xff),
-		SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+	sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC,
+		SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff),
+		SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff));
 
 	/* disable all clock gating */
 	writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
@@ -313,9 +310,7 @@ static const struct sst_adsp_memregion lp_region[] = {
 
 /* wild cat point ADSP mem regions */
 static const struct sst_adsp_memregion wpt_region[] = {
-	{0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
-	{0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
-	{0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+	{0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */
 	{0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
 };
 
@@ -339,26 +334,56 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
 	return 0;
 }
 
+struct sst_sram_shift {
+	u32 dev_id;	/* SST Device IDs  */
+	u32 iram_shift;
+	u32 dram_shift;
+};
+
+static const struct sst_sram_shift sram_shift[] = {
+	{SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
+	{SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
+};
 static u32 hsw_block_get_bit(struct sst_mem_block *block)
 {
-	u32 bit = 0, shift = 0;
+	u32 bit = 0, shift = 0, index;
+	struct sst_dsp *sst = block->dsp;
 
-	switch (block->type) {
-	case SST_MEM_DRAM:
-		shift = 16;
-		break;
-	case SST_MEM_IRAM:
-		shift = 6;
-		break;
-	default:
-		return 0;
+	for (index = 0; index < ARRAY_SIZE(sram_shift); index++) {
+		if (sram_shift[index].dev_id == sst->id)
+			break;
 	}
 
+	if (index < ARRAY_SIZE(sram_shift)) {
+		switch (block->type) {
+		case SST_MEM_DRAM:
+			shift = sram_shift[index].dram_shift;
+			break;
+		case SST_MEM_IRAM:
+			shift = sram_shift[index].iram_shift;
+			break;
+		default:
+			shift = 0;
+		}
+	} else
+		shift = 0;
+
 	bit = 1 << (block->index + shift);
 
 	return bit;
 }
 
+/*dummy read a SRAM block.*/
+static void sst_mem_block_dummy_read(struct sst_mem_block *block)
+{
+	u32 size;
+	u8 tmp_buf[4];
+	struct sst_dsp *sst = block->dsp;
+
+	size = block->size > 4 ? 4 : block->size;
+	memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size);
+}
+
 /* enable 32kB memory block - locks held by caller */
 static int hsw_block_enable(struct sst_mem_block *block)
 {
@@ -378,6 +403,8 @@ static int hsw_block_enable(struct sst_mem_block *block)
 	/* wait 18 DSP clock ticks */
 	udelay(10);
 
+	/*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/
+	sst_mem_block_dummy_read(block);
 	return 0;
 }
 
@@ -488,8 +515,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 		}
 	}
 
-	/* set default power gating mask */
-	writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+	/* set default power gating control, enable power gating control for all blocks. that is,
+	can't be accessed, please enable each block before accessing. */
+	writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
 
 	return 0;
 }
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 434236343ddf..b6291516dbbf 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready {
 	u32 inbox_size;
 	u32 outbox_size;
 	u32 fw_info_size;
-	u8 fw_info[1];
+	u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
 } __attribute__((packed));
 
 struct ipc_message {
@@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work)
 		return;
 	}
 
-	/* if the DSP is busy we will TX messages after IRQ */
+	/* if the DSP is busy, we will TX messages after IRQ.
+	 * also postpone if we are in the middle of procesing completion irq*/
 	ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
-	if (ipcx & SST_IPCX_BUSY) {
+	if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) {
 		spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
 		return;
 	}
@@ -502,6 +503,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg,
 		ipc_shim_dbg(hsw, "message timeout");
 
 		trace_ipc_error("error message timeout for", msg->header);
+		list_del(&msg->list);
 		ret = -ETIMEDOUT;
 	} else {
 
@@ -569,6 +571,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
 {
 	struct sst_hsw_ipc_fw_ready fw_ready;
 	u32 offset;
+	u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
+	char *tmp[5], *pinfo;
+	int i = 0;
 
 	offset = (header & 0x1FFFFFFF) << 3;
 
@@ -589,6 +594,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
 		fw_ready.inbox_offset, fw_ready.inbox_size);
 	dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
 		fw_ready.outbox_offset, fw_ready.outbox_size);
+	if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) {
+		fw_ready.fw_info[fw_ready.fw_info_size] = 0;
+		dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info);
+
+		/* log the FW version info got from the mailbox here. */
+		memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size);
+		pinfo = &fw_info[0];
+		for (i = 0; i < sizeof(tmp) / sizeof(char *); i++)
+			tmp[i] = strsep(&pinfo, " ");
+		dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - "
+			"version: %s.%s, build %s, source commit id: %s\n",
+			tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]);
+	}
 }
 
 static void hsw_notification_work(struct work_struct *work)
@@ -671,7 +689,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
 	switch (stream_msg) {
 	case IPC_STR_STAGE_MESSAGE:
 	case IPC_STR_NOTIFICATION:
+		break;
 	case IPC_STR_RESET:
+		trace_ipc_notification("stream reset", stream->reply.stream_hw_id);
 		break;
 	case IPC_STR_PAUSE:
 		stream->running = false;
@@ -762,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
 	}
 
 	/* update any stream states */
-	hsw_stream_update(hsw, msg);
+	if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE)
+		hsw_stream_update(hsw, msg);
 
 	/* wake up and return the error if we have waiters on this message ? */
 	list_del(&msg->list);
@@ -1628,7 +1649,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
 {
 	u32 header, state_;
-	int ret;
+	int ret, item;
 
 	header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
 	state_ = state;
@@ -1642,6 +1663,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 		return ret;
 	}
 
+	for (item = 0; item < dx->entries_no; item++) {
+		dev_dbg(hsw->dev,
+			"Item[%d] offset[%x] - size[%x] - source[%x]\n",
+			item, dx->mem_info[item].offset,
+			dx->mem_info[item].size,
+			dx->mem_info[item].source);
+	}
 	dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
 		dx->entries_no, state);
 
@@ -1775,8 +1803,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
 
 	/* get the FW version */
 	sst_hsw_fw_get_version(hsw, &version);
-	dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
-		version.type, version.major, version.minor, version.build);
 
 	/* get the globalmixer */
 	ret = sst_hsw_mixer_get_info(hsw);
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 058efb17c568..61bf6da4bb02 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = {
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME |
 				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE |
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
 				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= PAGE_SIZE,
 	.period_bytes_max	= (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE,
@@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
 		sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16);
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		bits = SST_HSW_DEPTH_24BIT;
+		bits = SST_HSW_DEPTH_32BIT;
+		sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24);
+		break;
+	case SNDRV_PCM_FORMAT_S8:
+		bits = SST_HSW_DEPTH_8BIT;
+		sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		bits = SST_HSW_DEPTH_32BIT;
 		sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32);
 		break;
 	default:
@@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
 }
 
 #define HSW_FORMATS \
-	(SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
-	 SNDRV_PCM_FMTBIT_S32_LE)
+	(SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
+	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
+	SNDRV_PCM_FMTBIT_S8)
 
 static struct snd_soc_dai_driver hsw_dais[] = {
 	{
@@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = {
 			.channels_min = 2,
 			.channels_max = 2,
 			.rates = SNDRV_PCM_RATE_48000,
-			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
 		},
 	},
 	{
@@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
 			.stream_name = "Loopback Capture",
 			.channels_min = 2,
 			.channels_max = 2,
-			.rates = SNDRV_PCM_RATE_8000_192000,
-			.formats = HSW_FORMATS,
+			.rates = SNDRV_PCM_RATE_48000,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
 		},
 	},
 	{
@@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
 			.stream_name = "Analog Capture",
 			.channels_min = 2,
 			.channels_max = 2,
-			.rates = SNDRV_PCM_RATE_8000_192000,
-			.formats = HSW_FORMATS,
+			.rates = SNDRV_PCM_RATE_48000,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
 		},
 	},
 };
diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h
index 8d482d76475a..4257263157cd 100644
--- a/sound/soc/intel/sst-mfld-dsp.h
+++ b/sound/soc/intel/sst-mfld-dsp.h
@@ -3,7 +3,7 @@
 /*
  *  sst_mfld_dsp.h - Intel SST Driver for audio engine
  *
- *  Copyright (C) 2008-12 Intel Corporation
+ *  Copyright (C) 2008-14 Intel Corporation
  *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
  *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  *
@@ -19,6 +19,142 @@
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */
 
+#define SST_MAX_BIN_BYTES 1024
+
+#define MAX_DBG_RW_BYTES 80
+#define MAX_NUM_SCATTER_BUFFERS 8
+#define MAX_LOOP_BACK_DWORDS 8
+/* IPC base address and mailbox, timestamp offsets */
+#define SST_MAILBOX_SIZE 0x0400
+#define SST_MAILBOX_SEND 0x0000
+#define SST_TIME_STAMP 0x1800
+#define SST_TIME_STAMP_MRFLD 0x800
+#define SST_RESERVED_OFFSET 0x1A00
+#define SST_SCU_LPE_MAILBOX 0x1000
+#define SST_LPE_SCU_MAILBOX 0x1400
+#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16)
+#define PROCESS_MSG 0x80
+
+/* Message ID's for IPC messages */
+/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */
+
+/* I2L Firmware/Codec Download msgs */
+#define IPC_IA_PREP_LIB_DNLD 0x01
+#define IPC_IA_LIB_DNLD_CMPLT 0x02
+#define IPC_IA_GET_FW_VERSION 0x04
+#define IPC_IA_GET_FW_BUILD_INF 0x05
+#define IPC_IA_GET_FW_INFO 0x06
+#define IPC_IA_GET_FW_CTXT 0x07
+#define IPC_IA_SET_FW_CTXT 0x08
+#define IPC_IA_PREPARE_SHUTDOWN 0x31
+/* I2L Codec Config/control msgs */
+#define IPC_PREP_D3 0x10
+#define IPC_IA_SET_CODEC_PARAMS 0x10
+#define IPC_IA_GET_CODEC_PARAMS 0x11
+#define IPC_IA_SET_PPP_PARAMS 0x12
+#define IPC_IA_GET_PPP_PARAMS 0x13
+#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA
+#define IPC_IA_ALG_PARAMS 0x1A
+#define IPC_IA_TUNING_PARAMS 0x1B
+#define IPC_IA_SET_RUNTIME_PARAMS 0x1C
+#define IPC_IA_SET_PARAMS 0x1
+#define IPC_IA_GET_PARAMS 0x2
+
+#define IPC_EFFECTS_CREATE 0xE
+#define IPC_EFFECTS_DESTROY 0xF
+
+/* I2L Stream config/control msgs */
+#define IPC_IA_ALLOC_STREAM_MRFLD 0x2
+#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */
+#define IPC_IA_FREE_STREAM_MRFLD 0x03
+#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */
+#define IPC_IA_SET_STREAM_PARAMS 0x22
+#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12
+#define IPC_IA_GET_STREAM_PARAMS 0x23
+#define IPC_IA_PAUSE_STREAM 0x24
+#define IPC_IA_PAUSE_STREAM_MRFLD 0x4
+#define IPC_IA_RESUME_STREAM 0x25
+#define IPC_IA_RESUME_STREAM_MRFLD 0x5
+#define IPC_IA_DROP_STREAM 0x26
+#define IPC_IA_DROP_STREAM_MRFLD 0x07
+#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */
+#define IPC_IA_DRAIN_STREAM_MRFLD 0x8
+#define IPC_IA_CONTROL_ROUTING 0x29
+#define IPC_IA_VTSV_UPDATE_MODULES 0x20
+#define IPC_IA_VTSV_DETECTED 0x21
+
+#define IPC_IA_START_STREAM_MRFLD 0X06
+#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */
+
+#define IPC_IA_SET_GAIN_MRFLD 0x21
+/* Debug msgs */
+#define IPC_IA_DBG_MEM_READ 0x40
+#define IPC_IA_DBG_MEM_WRITE 0x41
+#define IPC_IA_DBG_LOOP_BACK 0x42
+#define IPC_IA_DBG_LOG_ENABLE 0x45
+#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47
+
+/* L2I Firmware/Codec Download msgs */
+#define IPC_IA_FW_INIT_CMPLT 0x81
+#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01
+#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11
+
+/* L2I Codec Config/control msgs */
+#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */
+
+#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */
+#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */
+#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */
+#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */
+#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */
+#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */
+
+#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */
+/* L2S messages */
+#define IPC_SC_DDR_LINK_UP 0xC0
+#define IPC_SC_DDR_LINK_DOWN 0xC1
+#define IPC_SC_SET_LPECLK_REQ 0xC2
+#define IPC_SC_SSP_BIT_BANG 0xC3
+
+/* L2I Error reporting msgs */
+#define IPC_IA_MEM_ALLOC_FAIL 0xE0
+#define IPC_IA_PROC_ERR 0xE1 /* error in processing a
+					stream can be used by playback and
+					capture modules */
+
+/* L2I Debug msgs */
+#define IPC_IA_PRINT_STRING 0xF0
+
+/* Buffer under-run */
+#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B
+
+/* Mrfld specific defines:
+ * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR)
+ * received from FW, the format is:
+ *  - IPC High: pvt_id is set to zero. Always short message.
+ *  - msg_id is in lower 16-bits of IPC low payload.
+ *  - pipe_id is in higher 16-bits of IPC low payload for period_elapsed.
+ *  - error id is in higher 16-bits of IPC low payload for async errors.
+ */
+#define SST_ASYNC_DRV_ID 0
+
+/* Command Response or Acknowledge message to any IPC message will have
+ * same message ID and stream ID information which is sent.
+ * There is no specific Ack message ID. The data field is used as response
+ * meaning.
+ */
+enum ackData {
+	IPC_ACK_SUCCESS = 0,
+	IPC_ACK_FAILURE,
+};
+
+enum ipc_ia_msg_id {
+	IPC_CMD = 1,		/*!< Task Control message ID */
+	IPC_SET_PARAMS = 2,/*!< Task Set param message ID */
+	IPC_GET_PARAMS = 3,	/*!< Task Get param message ID */
+	IPC_INVALID = 0xFF,	/*!<Task Get param message ID */
+};
+
 enum sst_codec_types {
 	/*  AUDIO/MUSIC	CODEC Type Definitions */
 	SST_CODEC_TYPE_UNKNOWN = 0,
@@ -35,14 +171,157 @@ enum stream_type {
 	SST_STREAM_TYPE_MUSIC = 1,
 };
 
+enum sst_error_codes {
+	/* Error code,response to msgId: Description */
+	/* Common error codes */
+	SST_SUCCESS = 0,        /* Success */
+	SST_ERR_INVALID_STREAM_ID = 1,
+	SST_ERR_INVALID_MSG_ID = 2,
+	SST_ERR_INVALID_STREAM_OP = 3,
+	SST_ERR_INVALID_PARAMS = 4,
+	SST_ERR_INVALID_CODEC = 5,
+	SST_ERR_INVALID_MEDIA_TYPE = 6,
+	SST_ERR_STREAM_ERR = 7,
+
+	SST_ERR_STREAM_IN_USE = 15,
+};
+
+struct ipc_dsp_hdr {
+	u16 mod_index_id:8;		/*!< DSP Command ID specific to tasks */
+	u16 pipe_id:8;	/*!< instance of the module in the pipeline */
+	u16 mod_id;		/*!< Pipe_id */
+	u16 cmd_id;		/*!< Module ID = lpe_algo_types_t */
+	u16 length;		/*!< Length of the payload only */
+} __packed;
+
+union ipc_header_high {
+	struct {
+		u32  msg_id:8;	    /* Message ID - Max 256 Message Types */
+		u32  task_id:4;	    /* Task ID associated with this comand */
+		u32  drv_id:4;    /* Identifier for the driver to track*/
+		u32  rsvd1:8;	    /* Reserved */
+		u32  result:4;	    /* Reserved */
+		u32  res_rqd:1;	    /* Response rqd */
+		u32  large:1;	    /* Large Message if large = 1 */
+		u32  done:1;	    /* bit 30 - Done bit */
+		u32  busy:1;	    /* bit 31 - busy bit*/
+	} part;
+	u32 full;
+} __packed;
+/* IPC header */
+union ipc_header_mrfld {
+	struct {
+		u32 header_low_payload;
+		union ipc_header_high header_high;
+	} p;
+	u64 full;
+} __packed;
+/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/
+
+/* IPC Header */
+union ipc_header {
+	struct {
+		u32  msg_id:8; /* Message ID - Max 256 Message Types */
+		u32  str_id:5;
+		u32  large:1;	/* Large Message if large = 1 */
+		u32  reserved:2;	/* Reserved for future use */
+		u32  data:14;	/* Ack/Info for msg, size of msg in Mailbox */
+		u32  done:1; /* bit 30 */
+		u32  busy:1; /* bit 31 */
+	} part;
+	u32 full;
+} __packed;
+
+/* Firmware build info */
+struct sst_fw_build_info {
+	unsigned char  date[16]; /* Firmware build date */
+	unsigned char  time[16]; /* Firmware build time */
+} __packed;
+
+/* Firmware Version info */
+struct snd_sst_fw_version {
+	u8 build;	/* build number*/
+	u8 minor;	/* minor number*/
+	u8 major;	/* major number*/
+	u8 type;	/* build type */
+};
+
+struct ipc_header_fw_init {
+	struct snd_sst_fw_version fw_version;/* Firmware version details */
+	struct sst_fw_build_info build_info;
+	u16 result;	/* Fw init result */
+	u8 module_id; /* Module ID in case of error */
+	u8 debug_info; /* Debug info from Module ID in case of fail */
+} __packed;
+
+struct snd_sst_tstamp {
+	u64 ring_buffer_counter;	/* PB/CP: Bytes copied from/to DDR. */
+	u64 hardware_counter;	    /* PB/CP: Bytes DMAed to/from SSP. */
+	u64 frames_decoded;
+	u64 bytes_decoded;
+	u64 bytes_copied;
+	u32 sampling_frequency;
+	u32 channel_peak[8];
+} __packed;
+
+/* Stream type params struture for Alloc stream */
+struct snd_sst_str_type {
+	u8 codec_type;		/* Codec type */
+	u8 str_type;		/* 1 = voice 2 = music */
+	u8 operation;		/* Playback or Capture */
+	u8 protected_str;	/* 0=Non DRM, 1=DRM */
+	u8 time_slots;
+	u8 reserved;		/* Reserved */
+	u16 result;		/* Result used for acknowledgment */
+} __packed;
+
+/* Library info structure */
+struct module_info {
+	u32 lib_version;
+	u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/
+	u32 media_type;
+	u8  lib_name[12];
+	u32 lib_caps;
+	unsigned char  b_date[16]; /* Lib build date */
+	unsigned char  b_time[16]; /* Lib build time */
+} __packed;
+
+/* Library slot info */
+struct lib_slot_info {
+	u8  slot_num; /* 1 or 2 */
+	u8  reserved1;
+	u16 reserved2;
+	u32 iram_size; /* slot size in IRAM */
+	u32 dram_size; /* slot size in DRAM */
+	u32 iram_offset; /* starting offset of slot in IRAM */
+	u32 dram_offset; /* starting offset of slot in DRAM */
+} __packed;
+
+struct snd_ppp_mixer_params {
+	__u32			type; /*Type of the parameter */
+	__u32			size;
+	__u32			input_stream_bitmap; /*Input stream Bit Map*/
+} __packed;
+
+struct snd_sst_lib_download {
+	struct module_info lib_info; /* library info type, capabilities etc */
+	struct lib_slot_info slot_info; /* slot info to be downloaded */
+	u32 mod_entry_pt;
+};
+
+struct snd_sst_lib_download_info {
+	struct snd_sst_lib_download dload_lib;
+	u16 result;	/* Result used for acknowledgment */
+	u8 pvt_id; /* Private ID */
+	u8 reserved;  /* for alignment */
+};
 struct snd_pcm_params {
 	u8 num_chan;	/* 1=Mono, 2=Stereo */
 	u8 pcm_wd_sz;	/* 16/24 - bit*/
-	u32 reserved;	/* Bitrate in bits per second */
-	u32 sfreq;	/* Sampling rate in Hz */
-	u8 use_offload_path;
+	u8 use_offload_path;	/* 0-PCM using period elpased & ALSA interfaces
+				   1-PCM stream via compressed interface  */
 	u8 reserved2;
-	u16 reserved3;
+	u32 sfreq;    /* Sampling rate in Hz */
 	u8 channel_map[8];
 } __packed;
 
@@ -76,6 +355,7 @@ struct snd_aac_params {
 struct snd_wma_params {
 	u8  num_chan;	/* 1=Mono, 2=Stereo */
 	u8  pcm_wd_sz;	/* 16/24 - bit*/
+	u16 reserved1;
 	u32 brate;	/* Use the hard coded value. */
 	u32 sfreq;	/* Sampling freq eg. 8000, 441000, 48000 */
 	u32 channel_mask;  /* Channel Mask */
@@ -101,26 +381,153 @@ struct sst_address_info {
 };
 
 struct snd_sst_alloc_params_ext {
-	struct sst_address_info  ring_buf_info[8];
-	u8 sg_count;
-	u8 reserved;
-	u16 reserved2;
-	u32 frag_size;	/*Number of samples after which period elapsed
+	__u16 sg_count;
+	__u16 reserved;
+	__u32 frag_size;	/*Number of samples after which period elapsed
 				  message is sent valid only if path  = 0*/
-} __packed;
+	struct sst_address_info  ring_buf_info[8];
+};
 
 struct snd_sst_stream_params {
 	union snd_sst_codec_params uc;
 } __packed;
 
 struct snd_sst_params {
+	u32 result;
 	u32 stream_id;
 	u8 codec;
 	u8 ops;
 	u8 stream_type;
 	u8 device_type;
+	u8 task;
 	struct snd_sst_stream_params sparams;
 	struct snd_sst_alloc_params_ext aparams;
 };
 
+struct snd_sst_alloc_mrfld {
+	u16 codec_type;
+	u8 operation;
+	u8 sg_count;
+	struct sst_address_info ring_buf_info[8];
+	u32 frag_size;
+	u32 ts;
+	struct snd_sst_stream_params codec_params;
+} __packed;
+
+/* Alloc stream params structure */
+struct snd_sst_alloc_params {
+	struct snd_sst_str_type str_type;
+	struct snd_sst_stream_params stream_params;
+	struct snd_sst_alloc_params_ext alloc_params;
+} __packed;
+
+/* Alloc stream response message */
+struct snd_sst_alloc_response {
+	struct snd_sst_str_type str_type; /* Stream type for allocation */
+	struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */
+};
+
+/* Drop response */
+struct snd_sst_drop_response {
+	u32 result;
+	u32 bytes;
+};
+
+struct snd_sst_async_msg {
+	u32 msg_id; /* Async msg id */
+	u32 payload[0];
+};
+
+struct snd_sst_async_err_msg {
+	u32 fw_resp; /* Firmware Result */
+	u32 lib_resp; /*Library result */
+} __packed;
+
+struct snd_sst_vol {
+	u32	stream_id;
+	s32	volume;
+	u32	ramp_duration;
+	u32	ramp_type;		/* Ramp type, default=0 */
+};
+
+/* Gain library parameters for mrfld
+ * based on DSP command spec v0.82
+ */
+struct snd_sst_gain_v2 {
+	u16 gain_cell_num;  /* num of gain cells to modify*/
+	u8 cell_nbr_idx; /* instance index*/
+	u8 cell_path_idx; /* pipe-id */
+	u16 module_id; /*module id */
+	u16 left_cell_gain; /* left gain value in dB*/
+	u16 right_cell_gain; /* right gain value in dB*/
+	u16 gain_time_const; /* gain time constant*/
+} __packed;
+
+struct snd_sst_mute {
+	u32	stream_id;
+	u32	mute;
+};
+
+struct snd_sst_runtime_params {
+	u8 type;
+	u8 str_id;
+	u8 size;
+	u8 rsvd;
+	void *addr;
+} __packed;
+
+enum stream_param_type {
+	SST_SET_TIME_SLOT = 0,
+	SST_SET_CHANNEL_INFO = 1,
+	OTHERS = 2, /*reserved for future params*/
+};
+
+/* CSV Voice call routing structure */
+struct snd_sst_control_routing {
+	u8 control; /* 0=start, 1=Stop */
+	u8 reserved[3];	/* Reserved- for 32 bit alignment */
+};
+
+struct ipc_post {
+	struct list_head node;
+	union ipc_header header; /* driver specific */
+	bool is_large;
+	bool is_process_reply;
+	union ipc_header_mrfld mrfld_header;
+	char *mailbox_data;
+};
+
+struct snd_sst_ctxt_params {
+	u32 address; /* Physical Address in DDR where the context is stored */
+	u32 size; /* size of the context */
+};
+
+struct snd_sst_lpe_log_params {
+	u8 dbg_type;
+	u8 module_id;
+	u8 log_level;
+	u8 reserved;
+} __packed;
+
+enum snd_sst_bytes_type {
+	SND_SST_BYTES_SET = 0x1,
+	SND_SST_BYTES_GET = 0x2,
+};
+
+struct snd_sst_bytes_v2 {
+	u8 type;
+	u8 ipc_msg;
+	u8 block;
+	u8 task_id;
+	u8 pipe_id;
+	u8 rsvd;
+	u16 len;
+	char bytes[0];
+};
+
+#define MAX_VTSV_FILES 2
+struct snd_sst_vtsv_info {
+	struct sst_address_info vfiles[MAX_VTSV_FILES];
+} __packed;
+
 #endif /* __SST_MFLD_DSP_H__ */
diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c
index 02abd19fce1d..29c059ca19e8 100644
--- a/sound/soc/intel/sst-mfld-platform-compress.c
+++ b/sound/soc/intel/sst-mfld-platform-compress.c
@@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
 	int retval;
 	struct snd_sst_params str_params;
 	struct sst_compress_cb cb;
+	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
 
 	stream = cstream->runtime->private_data;
 	/* construct fw structure for this*/
 	memset(&str_params, 0, sizeof(str_params));
 
-	str_params.ops = STREAM_OPS_PLAYBACK;
-	str_params.stream_type = SST_STREAM_TYPE_MUSIC;
-	str_params.device_type = SND_SST_DEVICE_COMPRESS;
+	/* fill the device type and stream id to pass to SST driver */
+	retval = sst_fill_stream_params(cstream, ctx, &str_params, true);
+	pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval);
+	if (retval < 0)
+		return retval;
 
 	switch (params->codec.id) {
 	case SND_AUDIOCODEC_MP3: {
diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index 7c790f51d259..706212a6a68c 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -1,7 +1,7 @@
 /*
  *  sst_mfld_platform.c - Intel MID Platform driver
  *
- *  Copyright (C) 2010-2013 Intel Corp
+ *  Copyright (C) 2010-2014 Intel Corp
  *  Author: Vinod Koul <vinod.koul@intel.com>
  *  Author: Harsha Priya <priya.harsha@intel.com>
  *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -27,7 +27,9 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/compress_driver.h>
+#include <asm/platform_sst_audio.h>
 #include "sst-mfld-platform.h"
+#include "sst-atom-controls.h"
 
 struct sst_device *sst;
 static DEFINE_MUTEX(sst_lock);
@@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = {
 	.fifo_size = SST_FIFO_SIZE,
 };
 
+static struct sst_dev_stream_map dpcm_strm_map[] = {
+	{0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */
+	{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0},
+	{MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0},
+	{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
+};
+
 /* MFLD - MSIC */
 static struct snd_soc_dai_driver sst_platform_dai[] = {
 {
@@ -143,58 +152,142 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
 	return state;
 }
 
+static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
+				struct snd_sst_alloc_params_ext *alloc_param)
+{
+	unsigned int channels;
+	snd_pcm_uframes_t period_size;
+	ssize_t periodbytes;
+	ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+	u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+
+	channels = substream->runtime->channels;
+	period_size = substream->runtime->period_size;
+	periodbytes = samples_to_bytes(substream->runtime, period_size);
+	alloc_param->ring_buf_info[0].addr = buffer_addr;
+	alloc_param->ring_buf_info[0].size = buffer_bytes;
+	alloc_param->sg_count = 1;
+	alloc_param->reserved = 0;
+	alloc_param->frag_size = periodbytes * channels;
+
+}
 static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
-				struct sst_pcm_params *param)
+				struct snd_sst_stream_params *param)
 {
+	param->uc.pcm_params.num_chan = (u8) substream->runtime->channels;
+	param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits;
+	param->uc.pcm_params.sfreq = substream->runtime->rate;
+
+	/* PCM stream via ALSA interface */
+	param->uc.pcm_params.use_offload_path = 0;
+	param->uc.pcm_params.reserved2 = 0;
+	memset(param->uc.pcm_params.channel_map, 0, sizeof(u8));
 
-	param->num_chan = (u8) substream->runtime->channels;
-	param->pcm_wd_sz = substream->runtime->sample_bits;
-	param->reserved = 0;
-	param->sfreq = substream->runtime->rate;
-	param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream);
-	param->period_count = substream->runtime->period_size;
-	param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area);
-	pr_debug("period_cnt = %d\n", param->period_count);
-	pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz);
 }
 
-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
+static int sst_get_stream_mapping(int dev, int sdev, int dir,
+	struct sst_dev_stream_map *map, int size)
+{
+	int i;
+
+	if (map == NULL)
+		return -EINVAL;
+
+
+	/* index 0 is not used in stream map */
+	for (i = 1; i < size; i++) {
+		if ((map[i].dev_num == dev) && (map[i].direction == dir))
+			return i;
+	}
+	return 0;
+}
+
+int sst_fill_stream_params(void *substream,
+	const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress)
+{
+	int map_size;
+	int index;
+	struct sst_dev_stream_map *map;
+	struct snd_pcm_substream *pstream = NULL;
+	struct snd_compr_stream *cstream = NULL;
+
+	map = ctx->pdata->pdev_strm_map;
+	map_size = ctx->pdata->strm_map_size;
+
+	if (is_compress == true)
+		cstream = (struct snd_compr_stream *)substream;
+	else
+		pstream = (struct snd_pcm_substream *)substream;
+
+	str_params->stream_type = SST_STREAM_TYPE_MUSIC;
+
+	/* For pcm streams */
+	if (pstream) {
+		index = sst_get_stream_mapping(pstream->pcm->device,
+					  pstream->number, pstream->stream,
+					  map, map_size);
+		if (index <= 0)
+			return -EINVAL;
+
+		str_params->stream_id = index;
+		str_params->device_type = map[index].device_id;
+		str_params->task = map[index].task_id;
+
+		str_params->ops = (u8)pstream->stream;
+	}
+
+	if (cstream) {
+		index = sst_get_stream_mapping(cstream->device->device,
+					       0, cstream->direction,
+					       map, map_size);
+		if (index <= 0)
+			return -EINVAL;
+		str_params->stream_id = index;
+		str_params->device_type = map[index].device_id;
+		str_params->task = map[index].task_id;
+
+		str_params->ops = (u8)cstream->direction;
+	}
+	return 0;
+}
+
+static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
+		struct snd_soc_platform *platform)
 {
 	struct sst_runtime_stream *stream =
 			substream->runtime->private_data;
-	struct sst_pcm_params param = {0};
-	struct sst_stream_params str_params = {0};
-	int ret_val;
+	struct snd_sst_stream_params param = {{{0,},},};
+	struct snd_sst_params str_params = {0};
+	struct snd_sst_alloc_params_ext alloc_params = {0};
+	int ret_val = 0;
+	struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
 
 	/* set codec params and inform SST driver the same */
 	sst_fill_pcm_params(substream, &param);
+	sst_fill_alloc_params(substream, &alloc_params);
 	substream->runtime->dma_area = substream->dma_buffer.area;
 	str_params.sparams = param;
-	str_params.codec =  param.codec;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		str_params.ops = STREAM_OPS_PLAYBACK;
-		str_params.device_type = substream->pcm->device + 1;
-		pr_debug("Playbck stream,Device %d\n",
-					substream->pcm->device);
-	} else {
-		str_params.ops = STREAM_OPS_CAPTURE;
-		str_params.device_type = SND_SST_DEVICE_CAPTURE;
-		pr_debug("Capture stream,Device %d\n",
-					substream->pcm->device);
-	}
-	ret_val = stream->ops->open(&str_params);
-	pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
+	str_params.aparams = alloc_params;
+	str_params.codec = SST_CODEC_TYPE_PCM;
+
+	/* fill the device type and stream id to pass to SST driver */
+	ret_val = sst_fill_stream_params(substream, ctx, &str_params, false);
 	if (ret_val < 0)
 		return ret_val;
 
-	stream->stream_info.str_id = ret_val;
-	pr_debug("str id :  %d\n", stream->stream_info.str_id);
+	stream->stream_info.str_id = str_params.stream_id;
+
+	ret_val = stream->ops->open(&str_params);
+	if (ret_val <= 0)
+		return ret_val;
+
+
 	return ret_val;
 }
 
-static void sst_period_elapsed(void *mad_substream)
+static void sst_period_elapsed(void *arg)
 {
-	struct snd_pcm_substream *substream = mad_substream;
+	struct snd_pcm_substream *substream = arg;
 	struct sst_runtime_stream *stream;
 	int status;
 
@@ -218,7 +311,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 	pr_debug("setting buffer ptr param\n");
 	sst_set_stream_status(stream, SST_PLATFORM_INIT);
 	stream->stream_info.period_elapsed = sst_period_elapsed;
-	stream->stream_info.mad_substream = substream;
+	stream->stream_info.arg = substream;
 	stream->stream_info.buffer_ptr = 0;
 	stream->stream_info.sfreq = substream->runtime->rate;
 	ret_val = stream->ops->device_control(
@@ -230,19 +323,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 }
 /* end -- helper functions */
 
-static int sst_platform_open(struct snd_pcm_substream *substream)
+static int sst_media_open(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
 {
+	int ret_val = 0;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct sst_runtime_stream *stream;
-	int ret_val;
-
-	pr_debug("sst_platform_open called\n");
-
-	snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
-	ret_val = snd_pcm_hw_constraint_integer(runtime,
-						SNDRV_PCM_HW_PARAM_PERIODS);
-	if (ret_val < 0)
-		return ret_val;
 
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
@@ -251,50 +337,69 @@ static int sst_platform_open(struct snd_pcm_substream *substream)
 
 	/* get the sst ops */
 	mutex_lock(&sst_lock);
-	if (!sst) {
+	if (!sst ||
+	    !try_module_get(sst->dev->driver->owner)) {
 		pr_err("no device available to run\n");
-		mutex_unlock(&sst_lock);
-		kfree(stream);
-		return -ENODEV;
-	}
-	if (!try_module_get(sst->dev->driver->owner)) {
-		mutex_unlock(&sst_lock);
-		kfree(stream);
-		return -ENODEV;
+		ret_val = -ENODEV;
+		goto out_ops;
 	}
 	stream->ops = sst->ops;
 	mutex_unlock(&sst_lock);
 
 	stream->stream_info.str_id = 0;
-	sst_set_stream_status(stream, SST_PLATFORM_INIT);
-	stream->stream_info.mad_substream = substream;
+
+	stream->stream_info.arg = substream;
 	/* allocate memory for SST API set */
 	runtime->private_data = stream;
 
-	return 0;
+	/* Make sure, that the period size is always even */
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_PERIODS, 2);
+
+	return snd_pcm_hw_constraint_integer(runtime,
+			 SNDRV_PCM_HW_PARAM_PERIODS);
+out_ops:
+	kfree(stream);
+	mutex_unlock(&sst_lock);
+	return ret_val;
 }
 
-static int sst_platform_close(struct snd_pcm_substream *substream)
+static void sst_media_close(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
 {
 	struct sst_runtime_stream *stream;
 	int ret_val = 0, str_id;
 
-	pr_debug("sst_platform_close called\n");
 	stream = substream->runtime->private_data;
 	str_id = stream->stream_info.str_id;
 	if (str_id)
 		ret_val = stream->ops->close(str_id);
 	module_put(sst->dev->driver->owner);
 	kfree(stream);
-	return ret_val;
 }
 
-static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
+static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform,
+					       struct snd_pcm_substream *substream)
+{
+	struct sst_data *sst = snd_soc_platform_get_drvdata(platform);
+	struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	u32 str_id = stream->stream_info.str_id;
+	unsigned int pipe_id;
+	pipe_id = map[str_id].device_id;
+
+	pr_debug("%s: got pipe_id = %#x for str_id = %d\n",
+		 __func__, pipe_id, str_id);
+	return pipe_id;
+}
+
+static int sst_media_prepare(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
 {
 	struct sst_runtime_stream *stream;
 	int ret_val = 0, str_id;
 
-	pr_debug("sst_platform_pcm_prepare called\n");
 	stream = substream->runtime->private_data;
 	str_id = stream->stream_info.str_id;
 	if (stream->stream_info.str_id) {
@@ -303,8 +408,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
 		return ret_val;
 	}
 
-	ret_val = sst_platform_alloc_stream(substream);
-	if (ret_val < 0)
+	ret_val = sst_platform_alloc_stream(substream, dai->platform);
+	if (ret_val <= 0)
 		return ret_val;
 	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
 			"%d", stream->stream_info.str_id);
@@ -316,6 +421,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
 	return ret_val;
 }
 
+static int sst_media_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+	return 0;
+}
+
+static int sst_media_hw_free(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_soc_dai_ops sst_media_dai_ops = {
+	.startup = sst_media_open,
+	.shutdown = sst_media_close,
+	.prepare = sst_media_prepare,
+	.hw_params = sst_media_hw_params,
+	.hw_free = sst_media_hw_free,
+};
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime;
+
+	if (substream->pcm->internal)
+		return 0;
+
+	runtime = substream->runtime;
+	runtime->hw = sst_platform_pcm_hw;
+	return 0;
+}
+
 static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
 					int cmd)
 {
@@ -331,7 +471,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
 		pr_debug("sst: Trigger Start\n");
 		str_cmd = SST_SND_START;
 		status = SST_PLATFORM_RUNNING;
-		stream->stream_info.mad_substream = substream;
+		stream->stream_info.arg = substream;
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		pr_debug("sst: in stop\n");
@@ -377,32 +517,15 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer
 		pr_err("sst: error code = %d\n", ret_val);
 		return ret_val;
 	}
-	return stream->stream_info.buffer_ptr;
-}
-
-static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params)
-{
-	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
-	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-
-	return 0;
-}
-
-static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	return snd_pcm_lib_free_pages(substream);
+	substream->runtime->delay = str_info->pcm_delay;
+	return str_info->buffer_ptr;
 }
 
 static struct snd_pcm_ops sst_platform_ops = {
 	.open = sst_platform_open,
-	.close = sst_platform_close,
 	.ioctl = snd_pcm_lib_ioctl,
-	.prepare = sst_platform_pcm_prepare,
 	.trigger = sst_platform_pcm_trigger,
 	.pointer = sst_platform_pcm_pointer,
-	.hw_params = sst_platform_pcm_hw_params,
-	.hw_free = sst_platform_pcm_hw_free,
 };
 
 static void sst_pcm_free(struct snd_pcm *pcm)
@@ -413,15 +536,15 @@ static void sst_pcm_free(struct snd_pcm *pcm)
 
 static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int retval = 0;
 
-	pr_debug("sst_pcm_new called\n");
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
-			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+	if (dai->driver->playback.channels_min ||
+			dai->driver->capture.channels_min) {
 		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
 			SNDRV_DMA_TYPE_CONTINUOUS,
-			snd_dma_continuous_data(GFP_KERNEL),
+			snd_dma_continuous_data(GFP_DMA),
 			SST_MIN_BUFFER, SST_MAX_BUFFER);
 		if (retval) {
 			pr_err("dma buffer allocationf fail\n");
@@ -445,10 +568,28 @@ static const struct snd_soc_component_driver sst_component = {
 
 static int sst_platform_probe(struct platform_device *pdev)
 {
+	struct sst_data *drv;
 	int ret;
+	struct sst_platform_data *pdata;
+
+	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
+	if (drv == NULL) {
+		pr_err("kzalloc failed\n");
+		return -ENOMEM;
+	}
+
+	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+	if (pdata == NULL) {
+		pr_err("kzalloc failed for pdata\n");
+		return -ENOMEM;
+	}
+
+	pdata->pdev_strm_map = dpcm_strm_map;
+	pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map);
+	drv->pdata = pdata;
+	mutex_init(&drv->lock);
+	dev_set_drvdata(&pdev->dev, drv);
 
-	pr_debug("sst_platform_probe called\n");
-	sst = NULL;
 	ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
 	if (ret) {
 		pr_err("registering soc platform failed\n");
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index 6c5e7dc49e3c..6c6a42c08e24 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -39,9 +39,10 @@ extern struct sst_device *sst;
 
 struct pcm_stream_info {
 	int str_id;
-	void *mad_substream;
-	void (*period_elapsed) (void *mad_substream);
+	void *arg;
+	void (*period_elapsed) (void *arg);
 	unsigned long long buffer_ptr;
+	unsigned long long pcm_delay;
 	int sfreq;
 };
 
@@ -62,7 +63,9 @@ enum sst_controls {
 	SST_SND_BUFFER_POINTER =	0x05,
 	SST_SND_STREAM_INIT =		0x06,
 	SST_SND_START	 =		0x07,
-	SST_MAX_CONTROLS =		0x07,
+	SST_SET_BYTE_STREAM =           0x100A,
+	SST_GET_BYTE_STREAM =           0x100B,
+	SST_MAX_CONTROLS = SST_GET_BYTE_STREAM,
 };
 
 enum sst_stream_ops {
@@ -124,8 +127,9 @@ struct compress_sst_ops {
 };
 
 struct sst_ops {
-	int (*open) (struct sst_stream_params *str_param);
+	int (*open) (struct snd_sst_params *str_param);
 	int (*device_control) (int cmd, void *arg);
+	int (*set_generic_params)(enum sst_controls cmd, void *arg);
 	int (*close) (unsigned int str_id);
 };
 
@@ -143,10 +147,27 @@ struct sst_device {
 	char *name;
 	struct device *dev;
 	struct sst_ops *ops;
+	struct platform_device *pdev;
 	struct compress_sst_ops *compr_ops;
 };
 
+struct sst_data;
 void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
+int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
+			   struct snd_sst_params *str_params, bool is_compress);
+
+struct sst_algo_int_control_v2 {
+	struct soc_mixer_control mc;
+	u16 module_id; /* module identifieer */
+	u16 pipe_id; /* location info: pipe_id + instance_id */
+	u16 instance_id;
+	unsigned int value; /* Value received is stored here */
+};
+struct sst_data {
+	struct platform_device *pdev;
+	struct sst_platform_data *pdata;
+	struct mutex lock;
+};
 int sst_register_dsp(struct sst_device *sst);
 int sst_unregister_dsp(struct sst_device *sst);
 #endif