Merge "Revert "ASoC: msm: remove unused msm-compr-q6-v2"" into android-msm-sturgeon-3.10android-wear-7.1.1_r0.47

3 files changed, 1769 insertions, 1 deletions

diff --git a/sound/soc/msm/qdsp6v2/Makefile b/sound/soc/msm/qdsp6v2/Makefile
index 8f9e67eda54b..8e318cb1b0eb 100644
--- a/sound/soc/msm/qdsp6v2/Makefile
+++ b/sound/soc/msm/qdsp6v2/Makefile
@@ -1,5 +1,5 @@
 snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o \
-			msm-compress-q6-v2.o \
+			msm-compress-q6-v2.o msm-compr-q6-v2.o \
 			msm-pcm-lpa-v2.o \
 			msm-pcm-afe-v2.o msm-pcm-voip-v2.o \
 			msm-pcm-voice-v2.o msm-dai-q6-hdmi-v2.o \
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
new file mode 100644
index 000000000000..87523ab84c1b
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
@@ -0,0 +1,1732 @@
+/* Copyright (c) 2012-2014, The Linux Foundation. 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 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6asm-v2.h>
+#include <sound/pcm_params.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio_ion.h>
+
+#include <sound/timer.h>
+
+#include "msm-compr-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+#include "audio_ocmem.h"
+#include <sound/tlv.h>
+
+#define COMPRE_CAPTURE_NUM_PERIODS	16
+/* Allocate the worst case frame size for compressed audio */
+#define COMPRE_CAPTURE_HEADER_SIZE	(sizeof(struct snd_compr_audio_info))
+/* Changing period size to 4032. 4032 will make sure COMPRE_CAPTURE_PERIOD_SIZE
+ * is 4096 with meta data size of 64 and MAX_NUM_FRAMES_PER_BUFFER 1
+ */
+#define COMPRE_CAPTURE_MAX_FRAME_SIZE	(4032)
+#define COMPRE_CAPTURE_PERIOD_SIZE	((COMPRE_CAPTURE_MAX_FRAME_SIZE + \
+					  COMPRE_CAPTURE_HEADER_SIZE) * \
+					  MAX_NUM_FRAMES_PER_BUFFER)
+#define COMPRE_OUTPUT_METADATA_SIZE	(sizeof(struct output_meta_data_st))
+#define COMPRESSED_LR_VOL_MAX_STEPS	0x20002000
+
+#define MAX_AC3_PARAM_SIZE		(18*2*sizeof(int))
+#define AMR_WB_BAND_MODE 8
+#define AMR_WB_DTX_MODE 0
+
+
+const DECLARE_TLV_DB_LINEAR(compr_rx_vol_gain, 0,
+			    COMPRESSED_LR_VOL_MAX_STEPS);
+struct snd_msm {
+	atomic_t audio_ocmem_req;
+};
+static struct snd_msm compressed_audio;
+
+static struct audio_locks the_locks;
+
+static struct snd_pcm_hardware msm_compr_hardware_capture = {
+	.info =		 (SNDRV_PCM_INFO_MMAP |
+				SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_INTERLEAVED |
+				SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+	.formats =	      SNDRV_PCM_FMTBIT_S16_LE,
+	.rates =		SNDRV_PCM_RATE_8000_48000,
+	.rate_min =	     8000,
+	.rate_max =	     48000,
+	.channels_min =	 1,
+	.channels_max =	 8,
+	.buffer_bytes_max =
+		COMPRE_CAPTURE_PERIOD_SIZE * COMPRE_CAPTURE_NUM_PERIODS ,
+	.period_bytes_min =	COMPRE_CAPTURE_PERIOD_SIZE,
+	.period_bytes_max = COMPRE_CAPTURE_PERIOD_SIZE,
+	.periods_min =	  COMPRE_CAPTURE_NUM_PERIODS,
+	.periods_max =	  COMPRE_CAPTURE_NUM_PERIODS,
+	.fifo_size =	    0,
+};
+
+static struct snd_pcm_hardware msm_compr_hardware_playback = {
+	.info =		 (SNDRV_PCM_INFO_MMAP |
+				SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_INTERLEAVED |
+				SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+	.formats =	      SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+	.rates =		SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+	.rate_min =	     8000,
+	.rate_max =	     48000,
+	.channels_min =	 1,
+	.channels_max =	 8,
+	.buffer_bytes_max =     1024 * 1024,
+	.period_bytes_min =	128 * 1024,
+	.period_bytes_max =     256 * 1024,
+	.periods_min =	  4,
+	.periods_max =	  8,
+	.fifo_size =	    0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+/* Add supported codecs for compress capture path */
+static uint32_t supported_compr_capture_codecs[] = {
+	SND_AUDIOCODEC_AMRWB
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+	.count = ARRAY_SIZE(supported_sample_rates),
+	.list = supported_sample_rates,
+	.mask = 0,
+};
+
+static bool msm_compr_capture_codecs(uint32_t req_codec)
+{
+	int i;
+	pr_debug("%s req_codec:%d\n", __func__, req_codec);
+	if (req_codec == 0)
+		return false;
+	for (i = 0; i < ARRAY_SIZE(supported_compr_capture_codecs); i++) {
+		if (req_codec == supported_compr_capture_codecs[i])
+			return true;
+	}
+	return false;
+}
+
+static void compr_event_handler(uint32_t opcode,
+		uint32_t token, uint32_t *payload, void *priv)
+{
+	struct compr_audio *compr = priv;
+	struct msm_audio *prtd = &compr->prtd;
+	struct snd_pcm_substream *substream = prtd->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct audio_aio_write_param param;
+	struct audio_aio_read_param read_param;
+	struct audio_buffer *buf = NULL;
+	phys_addr_t temp;
+	struct output_meta_data_st output_meta_data;
+	uint32_t *ptrmem = (uint32_t *)payload;
+	int i = 0;
+	int time_stamp_flag = 0;
+	int buffer_length = 0;
+	int stop_playback = 0;
+
+	pr_debug("%s opcode =%08x\n", __func__, opcode);
+	switch (opcode) {
+	case ASM_DATA_EVENT_WRITE_DONE_V2: {
+		uint32_t *ptrmem = (uint32_t *)&param;
+		pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+		pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+		prtd->pcm_irq_pos += prtd->pcm_count;
+		if (atomic_read(&prtd->start))
+			snd_pcm_period_elapsed(substream);
+		else
+			if (substream->timer_running)
+				snd_timer_interrupt(substream->timer, 1);
+		atomic_inc(&prtd->out_count);
+		wake_up(&the_locks.write_wait);
+		if (!atomic_read(&prtd->start)) {
+			atomic_set(&prtd->pending_buffer, 1);
+			break;
+		} else
+			atomic_set(&prtd->pending_buffer, 0);
+
+		/*
+		 * check for underrun
+		 */
+		snd_pcm_stream_lock_irq(substream);
+		if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
+			runtime->render_flag |= SNDRV_RENDER_STOPPED;
+			stop_playback = 1;
+		}
+		snd_pcm_stream_unlock_irq(substream);
+
+		if (stop_playback) {
+			pr_err("underrun! render stopped\n");
+			break;
+		}
+
+		buf = prtd->audio_client->port[IN].buf;
+		pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
+				__func__, prtd->pcm_count, prtd->out_head);
+		temp = buf[0].phys + (prtd->out_head * prtd->pcm_count);
+		pr_debug("%s:writing buffer[%d] from 0x%pK\n",
+			__func__, prtd->out_head, &temp);
+
+		if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+			time_stamp_flag = SET_TIMESTAMP;
+		else
+			time_stamp_flag = NO_TIMESTAMP;
+		memcpy(&output_meta_data, (char *)(buf->data +
+			prtd->out_head * prtd->pcm_count),
+			COMPRE_OUTPUT_METADATA_SIZE);
+
+		buffer_length = output_meta_data.frame_size;
+		pr_debug("meta_data_length: %d, frame_length: %d\n",
+			 output_meta_data.meta_data_length,
+			 output_meta_data.frame_size);
+		pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
+			 output_meta_data.timestamp_msw,
+			 output_meta_data.timestamp_lsw);
+		if (buffer_length == 0) {
+			pr_debug("Recieved a zero length buffer-break out");
+			break;
+		}
+		param.paddr = temp + output_meta_data.meta_data_length;
+		param.len = buffer_length;
+		param.msw_ts = output_meta_data.timestamp_msw;
+		param.lsw_ts = output_meta_data.timestamp_lsw;
+		param.flags = time_stamp_flag;
+		param.uid = prtd->session_id;
+		for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
+					i++, ++ptrmem)
+			pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+		if (q6asm_async_write(prtd->audio_client,
+					&param) < 0)
+			pr_err("%s:q6asm_async_write failed\n",
+				__func__);
+		else
+			prtd->out_head =
+				(prtd->out_head + 1) & (runtime->periods - 1);
+		break;
+	}
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		pr_debug("ASM_DATA_CMDRSP_EOS\n");
+		if (atomic_read(&prtd->eos)) {
+			pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
+			prtd->cmd_ack = 1;
+			wake_up(&the_locks.eos_wait);
+			atomic_set(&prtd->eos, 0);
+		}
+		break;
+	case ASM_DATA_EVENT_READ_DONE_V2: {
+		pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+		pr_debug("buf = %pK, data = 0x%X, *data = %pK,\n"
+			 "prtd->pcm_irq_pos = %d\n",
+				prtd->audio_client->port[OUT].buf,
+			 *(uint32_t *)prtd->audio_client->port[OUT].buf->data,
+				prtd->audio_client->port[OUT].buf->data,
+				prtd->pcm_irq_pos);
+
+		memcpy(prtd->audio_client->port[OUT].buf->data +
+			   prtd->pcm_irq_pos, (ptrmem + READDONE_IDX_SIZE),
+			   COMPRE_CAPTURE_HEADER_SIZE);
+		pr_debug("buf = %pK, updated data = 0x%X, *data = %pK\n",
+				prtd->audio_client->port[OUT].buf,
+			*(uint32_t *)(prtd->audio_client->port[OUT].buf->data +
+				prtd->pcm_irq_pos),
+				prtd->audio_client->port[OUT].buf->data);
+		if (!atomic_read(&prtd->start))
+			break;
+		pr_debug("frame size=%d, buffer = 0x%X\n",
+				ptrmem[READDONE_IDX_SIZE],
+				ptrmem[READDONE_IDX_BUFADD_LSW]);
+		if (ptrmem[READDONE_IDX_SIZE] > COMPRE_CAPTURE_MAX_FRAME_SIZE) {
+			pr_err("Frame length exceeded the max length");
+			break;
+		}
+		buf = prtd->audio_client->port[OUT].buf;
+
+		pr_debug("pcm_irq_pos=%d, buf[0].phys = 0x%pK\n",
+				prtd->pcm_irq_pos, &buf[0].phys);
+		read_param.len = prtd->pcm_count - COMPRE_CAPTURE_HEADER_SIZE;
+		read_param.paddr = buf[0].phys +
+			prtd->pcm_irq_pos + COMPRE_CAPTURE_HEADER_SIZE;
+		prtd->pcm_irq_pos += prtd->pcm_count;
+
+		if (atomic_read(&prtd->start))
+			snd_pcm_period_elapsed(substream);
+
+		q6asm_async_read(prtd->audio_client, &read_param);
+		break;
+	}
+	case APR_BASIC_RSP_RESULT: {
+		switch (payload[0]) {
+		case ASM_SESSION_CMD_RUN_V2: {
+			if (substream->stream
+				!= SNDRV_PCM_STREAM_PLAYBACK) {
+				atomic_set(&prtd->start, 1);
+				break;
+			}
+			if (!atomic_read(&prtd->pending_buffer))
+				break;
+			pr_debug("%s: writing %d bytes of buffer[%d] to dsp\n",
+				__func__, prtd->pcm_count, prtd->out_head);
+			buf = prtd->audio_client->port[IN].buf;
+			pr_debug("%s: writing buffer[%d] from 0x%pK head %d count %d\n",
+				__func__, prtd->out_head, &buf[0].phys,
+				prtd->pcm_count, prtd->out_head);
+			if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+				time_stamp_flag = SET_TIMESTAMP;
+			else
+				time_stamp_flag = NO_TIMESTAMP;
+			memcpy(&output_meta_data, (char *)(buf->data +
+				prtd->out_head * prtd->pcm_count),
+				COMPRE_OUTPUT_METADATA_SIZE);
+			buffer_length = output_meta_data.frame_size;
+			pr_debug("meta_data_length: %d, frame_length: %d\n",
+				 output_meta_data.meta_data_length,
+				 output_meta_data.frame_size);
+			pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
+				 output_meta_data.timestamp_msw,
+				 output_meta_data.timestamp_lsw);
+			param.paddr = buf[prtd->out_head].phys
+					+ output_meta_data.meta_data_length;
+			param.len = buffer_length;
+			param.msw_ts = output_meta_data.timestamp_msw;
+			param.lsw_ts = output_meta_data.timestamp_lsw;
+			param.flags = time_stamp_flag;
+			param.uid = prtd->session_id;
+			param.metadata_len = COMPRE_OUTPUT_METADATA_SIZE;
+			if (q6asm_async_write(prtd->audio_client,
+						&param) < 0)
+				pr_err("%s:q6asm_async_write failed\n",
+					__func__);
+			else
+				prtd->out_head =
+					(prtd->out_head + 1)
+					& (runtime->periods - 1);
+			atomic_set(&prtd->pending_buffer, 0);
+		}
+			break;
+		case ASM_STREAM_CMD_FLUSH:
+			pr_debug("ASM_STREAM_CMD_FLUSH\n");
+			prtd->cmd_ack = 1;
+			wake_up(&the_locks.flush_wait);
+			break;
+		default:
+			break;
+		}
+		break;
+	}
+	default:
+		pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+		break;
+	}
+}
+
+static int msm_compr_send_ddp_cfg(struct audio_client *ac,
+					struct snd_dec_ddp *ddp)
+{
+	int i, rc;
+	pr_debug("%s\n", __func__);
+	for (i = 0; i < ddp->params_length/2; i++) {
+		rc = q6asm_ds1_set_endp_params(ac, ddp->params_id[i],
+						ddp->params_value[i]);
+		if (rc) {
+			pr_err("sending params_id: %d failed\n",
+				ddp->params_id[i]);
+			return rc;
+		}
+	}
+	return 0;
+}
+
+static int msm_compr_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr = runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	struct snd_pcm_hw_params *params;
+	struct asm_aac_cfg aac_cfg;
+	uint16_t bits_per_sample = 16;
+	int ret;
+
+	struct asm_softpause_params softpause = {
+		.enable = SOFT_PAUSE_ENABLE,
+		.period = SOFT_PAUSE_PERIOD,
+		.step = SOFT_PAUSE_STEP,
+		.rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+	};
+	struct asm_softvolume_params softvol = {
+		.period = SOFT_VOLUME_PERIOD,
+		.step = SOFT_VOLUME_STEP,
+		.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+	};
+
+	pr_debug("%s\n", __func__);
+
+	params = &soc_prtd->dpcm[substream->stream].hw_params;
+	if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
+		bits_per_sample = 24;
+
+	ret = q6asm_open_write_v2(prtd->audio_client,
+			compr->codec, bits_per_sample);
+	if (ret < 0) {
+		pr_err("%s: Session out open failed\n",
+				__func__);
+		return -ENOMEM;
+	}
+	msm_pcm_routing_reg_phy_stream(
+			soc_prtd->dai_link->be_id,
+			prtd->audio_client->perf_mode,
+			prtd->session_id,
+			substream->stream);
+	/*
+	 * the number of channels are required to call volume api
+	 * accoridngly. So, get channels from hw params
+	 */
+	if ((params_channels(params) > 0) &&
+			(params_periods(params) <= runtime->hw.channels_max))
+		prtd->channel_mode = params_channels(params);
+
+	ret = q6asm_set_softpause(prtd->audio_client, &softpause);
+	if (ret < 0)
+		pr_err("%s: Send SoftPause Param failed ret=%d\n",
+				__func__, ret);
+	ret = q6asm_set_softvolume(prtd->audio_client, &softvol);
+	if (ret < 0)
+		pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+				__func__, ret);
+
+	ret = q6asm_set_io_mode(prtd->audio_client,
+			(COMPRESSED_IO | ASYNC_IO_MODE));
+	if (ret < 0) {
+		pr_err("%s: Set IO mode failed\n", __func__);
+		return -ENOMEM;
+	}
+
+	prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+	prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+	prtd->pcm_irq_pos = 0;
+	/* rate and channels are sent to audio driver */
+	prtd->samp_rate = runtime->rate;
+	prtd->channel_mode = runtime->channels;
+	prtd->out_head = 0;
+	atomic_set(&prtd->out_count, runtime->periods);
+
+	if (prtd->enabled)
+		return 0;
+
+	switch (compr->info.codec_param.codec.id) {
+	case SND_AUDIOCODEC_MP3:
+		/* No media format block for mp3 */
+		break;
+	case SND_AUDIOCODEC_AAC:
+		pr_debug("%s: SND_AUDIOCODEC_AAC\n", __func__);
+		memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
+		aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
+		aac_cfg.format = 0x03;
+		aac_cfg.ch_cfg = runtime->channels;
+		aac_cfg.sample_rate =  runtime->rate;
+		ret = q6asm_media_format_block_aac(prtd->audio_client,
+					&aac_cfg);
+		if (ret < 0)
+			pr_err("%s: CMD Format block failed\n", __func__);
+		break;
+	case SND_AUDIOCODEC_AC3: {
+		struct snd_dec_ddp *ddp =
+				&compr->info.codec_param.codec.options.ddp;
+		pr_debug("%s: SND_AUDIOCODEC_AC3\n", __func__);
+		ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
+		if (ret < 0)
+			pr_err("%s: DDP CMD CFG failed\n", __func__);
+		break;
+	}
+	case SND_AUDIOCODEC_EAC3: {
+		struct snd_dec_ddp *ddp =
+				&compr->info.codec_param.codec.options.ddp;
+		pr_debug("%s: SND_AUDIOCODEC_EAC3\n", __func__);
+		ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
+		if (ret < 0)
+			pr_err("%s: DDP CMD CFG failed\n", __func__);
+		break;
+	}
+	default:
+		return -EINVAL;
+	}
+
+	prtd->enabled = 1;
+	prtd->cmd_ack = 0;
+	prtd->cmd_interrupt = 0;
+
+	return 0;
+}
+
+static int msm_compr_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	struct audio_buffer *buf = prtd->audio_client->port[OUT].buf;
+	struct snd_codec *codec = &compr->info.codec_param.codec;
+	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct audio_aio_read_param read_param;
+	uint16_t bits_per_sample = 16;
+	int ret = 0;
+	int i;
+
+	prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+	prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+	prtd->pcm_irq_pos = 0;
+
+	if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
+		bits_per_sample = 24;
+
+	if (!msm_compr_capture_codecs(
+				compr->info.codec_param.codec.id)) {
+		/*
+		 * request codec invalid or not supported,
+		 * use default compress format
+		 */
+		compr->info.codec_param.codec.id =
+			SND_AUDIOCODEC_AMRWB;
+	}
+	switch (compr->info.codec_param.codec.id) {
+	case SND_AUDIOCODEC_AMRWB:
+		pr_debug("q6asm_open_read(FORMAT_AMRWB)\n");
+		ret = q6asm_open_read(prtd->audio_client,
+				FORMAT_AMRWB);
+		if (ret < 0) {
+			pr_err("%s: compressed Session out open failed\n",
+					__func__);
+			return -ENOMEM;
+		}
+		pr_debug("msm_pcm_routing_reg_phy_stream\n");
+		msm_pcm_routing_reg_phy_stream(
+				soc_prtd->dai_link->be_id,
+				prtd->audio_client->perf_mode,
+				prtd->session_id, substream->stream);
+		break;
+	default:
+		pr_debug("q6asm_open_read_compressed(COMPRESSED_META_DATA_MODE)\n");
+		/*
+		   ret = q6asm_open_read_compressed(prtd->audio_client,
+		   MAX_NUM_FRAMES_PER_BUFFER,
+		   COMPRESSED_META_DATA_MODE);
+		 */
+			ret = -EINVAL;
+			break;
+	}
+
+	if (ret < 0) {
+		pr_err("%s: compressed Session out open failed\n",
+				__func__);
+		return -ENOMEM;
+	}
+
+	ret = q6asm_set_io_mode(prtd->audio_client,
+		(COMPRESSED_IO | ASYNC_IO_MODE));
+		if (ret < 0) {
+			pr_err("%s: Set IO mode failed\n", __func__);
+				return -ENOMEM;
+		}
+
+	if (!msm_compr_capture_codecs(codec->id)) {
+		/*
+		 * request codec invalid or not supported,
+		 * use default compress format
+		 */
+		codec->id = SND_AUDIOCODEC_AMRWB;
+	}
+	/* rate and channels are sent to audio driver */
+	prtd->samp_rate = runtime->rate;
+	prtd->channel_mode = runtime->channels;
+
+	if (prtd->enabled)
+		return ret;
+	read_param.len = prtd->pcm_count;
+
+	switch (codec->id) {
+	case SND_AUDIOCODEC_AMRWB:
+		pr_debug("SND_AUDIOCODEC_AMRWB\n");
+		ret = q6asm_enc_cfg_blk_amrwb(prtd->audio_client,
+			MAX_NUM_FRAMES_PER_BUFFER,
+			/*
+			 * use fixed band mode and dtx mode
+			 * band mode - 23.85 kbps
+			 */
+			AMR_WB_BAND_MODE,
+			/* dtx mode - disable */
+			AMR_WB_DTX_MODE);
+		if (ret < 0)
+			pr_err("%s: CMD Format block" \
+				"failed: %d\n", __func__, ret);
+		break;
+	default:
+		pr_debug("No config for codec %d\n", codec->id);
+	}
+	pr_debug("%s: Samp_rate = %d, Channel = %d, pcm_size = %d,\n"
+			 "pcm_count = %d, periods = %d\n",
+			 __func__, prtd->samp_rate, prtd->channel_mode,
+			 prtd->pcm_size, prtd->pcm_count, runtime->periods);
+
+	for (i = 0; i < runtime->periods; i++) {
+		read_param.uid = i;
+		switch (codec->id) {
+		case SND_AUDIOCODEC_AMRWB:
+			read_param.len = prtd->pcm_count
+					- COMPRE_CAPTURE_HEADER_SIZE;
+			read_param.paddr = buf[i].phys
+					+ COMPRE_CAPTURE_HEADER_SIZE;
+			pr_debug("Push buffer [%d] to DSP, "\
+					"paddr: %pK, vaddr: %pK\n",
+					i, &read_param.paddr,
+					buf[i].data);
+			q6asm_async_read(prtd->audio_client, &read_param);
+			break;
+		default:
+			read_param.paddr = buf[i].phys;
+			/*q6asm_async_read_compressed(prtd->audio_client,
+				&read_param);*/
+			pr_debug("%s: To add support for read compressed\n",
+								__func__);
+			ret = -EINVAL;
+			break;
+		}
+	}
+	prtd->periods = runtime->periods;
+
+	prtd->enabled = 1;
+
+	return ret;
+}
+
+static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	int ret = 0;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+
+	pr_debug("%s\n", __func__);
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		prtd->pcm_irq_pos = 0;
+
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			if (!msm_compr_capture_codecs(
+				compr->info.codec_param.codec.id)) {
+				/*
+				 * request codec invalid or not supported,
+				 * use default compress format
+				 */
+				compr->info.codec_param.codec.id =
+				SND_AUDIOCODEC_AMRWB;
+			}
+			switch (compr->info.codec_param.codec.id) {
+			case SND_AUDIOCODEC_AMRWB:
+				break;
+			default:
+				msm_pcm_routing_reg_psthr_stream(
+					soc_prtd->dai_link->be_id,
+					prtd->session_id, substream->stream);
+				break;
+			}
+		}
+		atomic_set(&prtd->pending_buffer, 1);
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		pr_debug("%s: Trigger start\n", __func__);
+		q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+		atomic_set(&prtd->start, 1);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			switch (compr->info.codec_param.codec.id) {
+			case SND_AUDIOCODEC_AMRWB:
+				break;
+			default:
+				msm_pcm_routing_reg_psthr_stream(
+					soc_prtd->dai_link->be_id,
+					prtd->session_id, substream->stream);
+				break;
+			}
+		}
+		atomic_set(&prtd->start, 0);
+		runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+		q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+		atomic_set(&prtd->start, 0);
+		runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static void populate_codec_list(struct compr_audio *compr,
+		struct snd_pcm_runtime *runtime)
+{
+	pr_debug("%s\n", __func__);
+	/* MP3 Block */
+	compr->info.compr_cap.num_codecs = 5;
+	compr->info.compr_cap.min_fragment_size = runtime->hw.period_bytes_min;
+	compr->info.compr_cap.max_fragment_size = runtime->hw.period_bytes_max;
+	compr->info.compr_cap.min_fragments = runtime->hw.periods_min;
+	compr->info.compr_cap.max_fragments = runtime->hw.periods_max;
+	compr->info.compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
+	compr->info.compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
+	compr->info.compr_cap.codecs[2] = SND_AUDIOCODEC_AC3;
+	compr->info.compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3;
+	compr->info.compr_cap.codecs[4] = SND_AUDIOCODEC_AMRWB;
+	/* Add new codecs here */
+}
+
+static int msm_compr_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr;
+	struct msm_audio *prtd;
+	int ret = 0;
+
+	pr_debug("%s\n", __func__);
+	compr = kzalloc(sizeof(struct compr_audio), GFP_KERNEL);
+	if (compr == NULL) {
+		pr_err("Failed to allocate memory for msm_audio\n");
+		return -ENOMEM;
+	}
+	prtd = &compr->prtd;
+	prtd->substream = substream;
+	runtime->render_flag = SNDRV_DMA_MODE;
+	prtd->audio_client = q6asm_audio_client_alloc(
+				(app_cb)compr_event_handler, compr);
+	if (!prtd->audio_client) {
+		pr_info("%s: Could not allocate memory\n", __func__);
+		kfree(prtd);
+		return -ENOMEM;
+	}
+
+	prtd->audio_client->perf_mode = false;
+	pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+	prtd->session_id = prtd->audio_client->session;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		runtime->hw = msm_compr_hardware_playback;
+		prtd->cmd_ack = 1;
+	} else {
+		runtime->hw = msm_compr_hardware_capture;
+	}
+
+
+	ret = snd_pcm_hw_constraint_list(runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_sample_rates);
+	if (ret < 0)
+		pr_info("snd_pcm_hw_constraint_list failed\n");
+	/* Ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+			    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+	prtd->dsp_cnt = 0;
+	atomic_set(&prtd->pending_buffer, 1);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		compr->codec = FORMAT_MP3;
+	populate_codec_list(compr, runtime);
+	runtime->private_data = compr;
+	atomic_set(&prtd->eos, 0);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (!atomic_cmpxchg(&compressed_audio.audio_ocmem_req, 0, 1))
+			audio_ocmem_process_req(AUDIO, true);
+		else
+			atomic_inc(&compressed_audio.audio_ocmem_req);
+		pr_debug("%s: req: %d\n", __func__,
+			atomic_read(&compressed_audio.audio_ocmem_req));
+	}
+	return 0;
+}
+
+static int compressed_set_volume(struct msm_audio *prtd, uint32_t volume)
+{
+	int rc = 0;
+	int avg_vol = 0;
+	int lgain = (volume >> 16) & 0xFFFF;
+	int rgain = volume & 0xFFFF;
+	if (prtd && prtd->audio_client) {
+		pr_debug("%s: channels %d volume 0x%x\n", __func__,
+			prtd->channel_mode, volume);
+		if ((prtd->channel_mode == 2) &&
+			(lgain != rgain)) {
+			pr_debug("%s: call q6asm_set_lrgain\n", __func__);
+			rc = q6asm_set_lrgain(prtd->audio_client, lgain, rgain);
+		} else {
+			avg_vol = (lgain + rgain)/2;
+			pr_debug("%s: call q6asm_set_volume\n", __func__);
+			rc = q6asm_set_volume(prtd->audio_client, avg_vol);
+		}
+		if (rc < 0) {
+			pr_err("%s: Send Volume command failed rc=%d\n",
+				__func__, rc);
+		}
+	}
+	return rc;
+}
+
+static int msm_compr_playback_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	int dir = 0;
+
+	pr_debug("%s\n", __func__);
+
+	dir = IN;
+	atomic_set(&prtd->pending_buffer, 0);
+
+	if (atomic_read(&compressed_audio.audio_ocmem_req) > 1)
+		atomic_dec(&compressed_audio.audio_ocmem_req);
+	else if (atomic_cmpxchg(&compressed_audio.audio_ocmem_req, 1, 0))
+		audio_ocmem_process_req(AUDIO, false);
+
+	pr_debug("%s: req: %d\n", __func__,
+		atomic_read(&compressed_audio.audio_ocmem_req));
+	prtd->pcm_irq_pos = 0;
+	q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+	q6asm_audio_client_buf_free_contiguous(dir,
+				prtd->audio_client);
+		msm_pcm_routing_dereg_phy_stream(
+			soc_prtd->dai_link->be_id,
+			SNDRV_PCM_STREAM_PLAYBACK);
+	q6asm_audio_client_free(prtd->audio_client);
+	kfree(prtd);
+	return 0;
+}
+
+static int msm_compr_capture_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	int dir = OUT;
+
+	pr_debug("%s\n", __func__);
+	atomic_set(&prtd->pending_buffer, 0);
+	q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+	q6asm_audio_client_buf_free_contiguous(dir,
+				prtd->audio_client);
+	msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+				SNDRV_PCM_STREAM_CAPTURE);
+	q6asm_audio_client_free(prtd->audio_client);
+	kfree(prtd);
+	return 0;
+}
+
+static int msm_compr_close(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ret = msm_compr_playback_close(substream);
+	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		ret = msm_compr_capture_close(substream);
+	return ret;
+}
+
+static int msm_compr_prepare(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ret = msm_compr_playback_prepare(substream);
+	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		ret = msm_compr_capture_prepare(substream);
+	return ret;
+}
+
+static snd_pcm_uframes_t msm_compr_pointer(struct snd_pcm_substream *substream)
+{
+
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+
+	if (prtd->pcm_irq_pos >= prtd->pcm_size)
+		prtd->pcm_irq_pos = 0;
+
+	pr_debug("%s: pcm_irq_pos = %d, pcm_size = %d, sample_bits = %d,\n"
+			 "frame_bits = %d\n", __func__, prtd->pcm_irq_pos,
+			 prtd->pcm_size, runtime->sample_bits,
+			 runtime->frame_bits);
+	return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_compr_mmap(struct snd_pcm_substream *substream,
+				struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct msm_audio *prtd = runtime->private_data;
+	struct audio_client *ac = prtd->audio_client;
+	struct audio_port_data *apd = ac->port;
+	struct audio_buffer *ab;
+	int dir = -1;
+
+	prtd->mmap_flag = 1;
+	runtime->render_flag = SNDRV_NON_DMA_MODE;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = IN;
+	else
+		dir = OUT;
+	ab = &(apd[dir].buf[0]);
+
+	return msm_audio_ion_mmap(ab, vma);
+}
+
+static int msm_compr_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+	struct audio_buffer *buf;
+	int dir, ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = IN;
+	else
+		dir = OUT;
+	/* Modifying kernel hardware params based on userspace config */
+	if (params_periods(params) > 0 &&
+		(params_periods(params) != runtime->hw.periods_max)) {
+		runtime->hw.periods_max = params_periods(params);
+	}
+	if (params_period_bytes(params) > 0 &&
+		(params_period_bytes(params) != runtime->hw.period_bytes_min)) {
+		runtime->hw.period_bytes_min = params_period_bytes(params);
+	}
+	runtime->hw.buffer_bytes_max =
+			runtime->hw.period_bytes_min * runtime->hw.periods_max;
+	pr_debug("allocate %zd buffers each of size %d\n",
+		runtime->hw.period_bytes_min,
+		runtime->hw.periods_max);
+	ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+			prtd->audio_client,
+			runtime->hw.period_bytes_min,
+			runtime->hw.periods_max);
+	if (ret < 0) {
+		pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+						ret);
+		return -ENOMEM;
+	}
+	buf = prtd->audio_client->port[dir].buf;
+
+	dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	dma_buf->dev.dev = substream->pcm->card->dev;
+	dma_buf->private_data = NULL;
+	dma_buf->area = buf[0].data;
+	dma_buf->addr =  buf[0].phys;
+	dma_buf->bytes = runtime->hw.buffer_bytes_max;
+
+	pr_debug("%s: buf[%pK]dma_buf->area[%pK]dma_buf->addr[%pK]\n"
+		 "dma_buf->bytes[%zd]\n", __func__,
+		 (void *)buf, (void *)dma_buf->area,
+		 &dma_buf->addr, dma_buf->bytes);
+	if (!dma_buf->area)
+		return -ENOMEM;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	return 0;
+}
+
+static int msm_compr_ioctl_shared(struct snd_pcm_substream *substream,
+		unsigned int cmd, void *arg)
+{
+	int rc = 0;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	uint64_t timestamp;
+	uint64_t temp;
+
+	switch (cmd) {
+	case SNDRV_COMPRESS_TSTAMP: {
+		struct snd_compr_tstamp *tstamp;
+		pr_debug("SNDRV_COMPRESS_TSTAMP\n");
+		tstamp = arg;
+		memset(tstamp, 0x0, sizeof(*tstamp));
+		rc = q6asm_get_session_time(prtd->audio_client, &timestamp);
+		if (rc < 0) {
+			pr_err("%s: Get Session Time return value =%lld\n",
+				__func__, timestamp);
+			return -EAGAIN;
+		}
+		temp = (timestamp * 2 * runtime->channels);
+		temp = temp * (runtime->rate/1000);
+		temp = div_u64(temp, 1000);
+		tstamp->sampling_rate = runtime->rate;
+		tstamp->timestamp = timestamp;
+		pr_debug("%s: bytes_consumed:,timestamp = %lld,\n",
+						__func__,
+			tstamp->timestamp);
+		return 0;
+	}
+	case SNDRV_COMPRESS_GET_CAPS: {
+		struct snd_compr_caps *caps;
+		caps = arg;
+		memset(caps, 0, sizeof(*caps));
+		pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
+		memcpy(caps, &compr->info.compr_cap, sizeof(*caps));
+		return 0;
+	}
+	case SNDRV_COMPRESS_SET_PARAMS:
+		pr_debug("SNDRV_COMPRESS_SET_PARAMS:\n");
+		memcpy(&compr->info.codec_param, (void *) arg,
+			sizeof(struct snd_compr_params));
+		switch (compr->info.codec_param.codec.id) {
+		case SND_AUDIOCODEC_MP3:
+			/* For MP3 we dont need any other parameter */
+			pr_debug("SND_AUDIOCODEC_MP3\n");
+			compr->codec = FORMAT_MP3;
+			break;
+		case SND_AUDIOCODEC_AAC:
+			pr_debug("SND_AUDIOCODEC_AAC\n");
+			compr->codec = FORMAT_MPEG4_AAC;
+			break;
+		case SND_AUDIOCODEC_AC3: {
+			char params_value[MAX_AC3_PARAM_SIZE];
+			int *params_value_data = (int *)params_value;
+			/* 36 is the max param length for ddp */
+			int i;
+			struct snd_dec_ddp *ddp =
+				&compr->info.codec_param.codec.options.ddp;
+			uint32_t params_length = 0;
+			memset(params_value, 0, MAX_AC3_PARAM_SIZE);
+			/* check integer overflow */
+			if (ddp->params_length > UINT_MAX/sizeof(int)) {
+				pr_err("%s: Integer overflow ddp->params_length %d\n",
+				__func__, ddp->params_length);
+				return -EINVAL;
+			}
+			params_length = ddp->params_length*sizeof(int);
+			if (params_length > MAX_AC3_PARAM_SIZE) {
+				/*MAX is 36*sizeof(int) this should not happen*/
+				pr_err("%s: params_length(%d) is greater than %zd\n",
+				__func__, params_length, MAX_AC3_PARAM_SIZE);
+				return -EINVAL;
+			}
+			pr_debug("SND_AUDIOCODEC_AC3\n");
+			compr->codec = FORMAT_AC3;
+			if (copy_from_user(params_value, (void *)ddp->params,
+					params_length))
+				pr_err("%s: copy ddp params value, size=%d\n",
+					__func__, params_length);
+			pr_debug("params_length: %d\n", ddp->params_length);
+			for (i = 0; i < params_length/sizeof(int); i++)
+				pr_debug("params_value[%d]: %x\n", i,
+					params_value_data[i]);
+			for (i = 0; i < ddp->params_length/2; i++) {
+				ddp->params_id[i] = params_value_data[2*i];
+				ddp->params_value[i] = params_value_data[2*i+1];
+			}
+			if (atomic_read(&prtd->start)) {
+				rc = msm_compr_send_ddp_cfg(prtd->audio_client,
+								ddp);
+				if (rc < 0)
+					pr_err("%s: DDP CMD CFG failed\n",
+						__func__);
+			}
+			break;
+		}
+		case SND_AUDIOCODEC_EAC3: {
+			char params_value[MAX_AC3_PARAM_SIZE];
+			int *params_value_data = (int *)params_value;
+			/* 36 is the max param length for ddp */
+			int i;
+			struct snd_dec_ddp *ddp =
+				&compr->info.codec_param.codec.options.ddp;
+			uint32_t params_length = 0;
+			memset(params_value, 0, MAX_AC3_PARAM_SIZE);
+			/* check integer overflow */
+			if (ddp->params_length > UINT_MAX/sizeof(int)) {
+				pr_err("%s: Integer overflow ddp->params_length %d\n",
+				__func__, ddp->params_length);
+				return -EINVAL;
+			}
+			if (params_length > MAX_AC3_PARAM_SIZE) {
+				/*MAX is 36*sizeof(int) this should not happen*/
+				pr_err("%s: params_length(%d) is greater than %zd\n",
+				__func__, params_length, MAX_AC3_PARAM_SIZE);
+				return -EINVAL;
+			}
+			pr_debug("SND_AUDIOCODEC_EAC3\n");
+			compr->codec = FORMAT_EAC3;
+			if (copy_from_user(params_value, (void *)ddp->params,
+					params_length))
+				pr_err("%s: copy ddp params value, size=%d\n",
+					__func__, params_length);
+			pr_debug("params_length: %d\n", ddp->params_length);
+			for (i = 0; i < ddp->params_length; i++)
+				pr_debug("params_value[%d]: %x\n", i,
+					params_value_data[i]);
+			for (i = 0; i < ddp->params_length/2; i++) {
+				ddp->params_id[i] = params_value_data[2*i];
+				ddp->params_value[i] = params_value_data[2*i+1];
+			}
+			if (atomic_read(&prtd->start)) {
+				rc = msm_compr_send_ddp_cfg(prtd->audio_client,
+								ddp);
+				if (rc < 0)
+					pr_err("%s: DDP CMD CFG failed\n",
+						__func__);
+			}
+			break;
+		}
+		default:
+			pr_debug("FORMAT_LINEAR_PCM\n");
+			compr->codec = FORMAT_LINEAR_PCM;
+			break;
+		}
+		return 0;
+	case SNDRV_PCM_IOCTL1_RESET:
+		pr_debug("SNDRV_PCM_IOCTL1_RESET\n");
+		/* Flush only when session is started during CAPTURE,
+		   while PLAYBACK has no such restriction. */
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+			  (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+						atomic_read(&prtd->start))) {
+			if (atomic_read(&prtd->eos)) {
+				prtd->cmd_interrupt = 1;
+				wake_up(&the_locks.eos_wait);
+				atomic_set(&prtd->eos, 0);
+			}
+
+			/* A unlikely race condition possible with FLUSH
+			   DRAIN if ack is set by flush and reset by drain */
+			prtd->cmd_ack = 0;
+			rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+			if (rc < 0) {
+				pr_err("%s: flush cmd failed rc=%d\n",
+					__func__, rc);
+				return rc;
+			}
+			rc = wait_event_timeout(the_locks.flush_wait,
+				prtd->cmd_ack, 5 * HZ);
+			if (!rc)
+				pr_err("Flush cmd timeout\n");
+			prtd->pcm_irq_pos = 0;
+		}
+		break;
+	case SNDRV_COMPRESS_DRAIN:
+		pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__);
+		if (atomic_read(&prtd->pending_buffer)) {
+			pr_debug("%s: no pending writes, drain would block\n",
+			 __func__);
+			return -EWOULDBLOCK;
+		}
+
+		atomic_set(&prtd->eos, 1);
+		atomic_set(&prtd->pending_buffer, 0);
+		prtd->cmd_ack = 0;
+		q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+		/* Wait indefinitely for  DRAIN. Flush can also signal this*/
+		rc = wait_event_interruptible(the_locks.eos_wait,
+			(prtd->cmd_ack || prtd->cmd_interrupt));
+
+		if (rc < 0)
+			pr_err("EOS cmd interrupted\n");
+		pr_debug("%s: SNDRV_COMPRESS_DRAIN  out of wait\n", __func__);
+
+		if (prtd->cmd_interrupt)
+			rc = -EINTR;
+
+		prtd->cmd_interrupt = 0;
+		return rc;
+	default:
+		break;
+	}
+	return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+#ifdef CONFIG_COMPAT
+struct snd_enc_wma32 {
+	u32 super_block_align; /* WMA Type-specific data */
+	u32 encodeopt1;
+	u32 encodeopt2;
+};
+
+struct snd_enc_vorbis32 {
+	s32 quality;
+	u32 managed;
+	u32 max_bit_rate;
+	u32 min_bit_rate;
+	u32 downmix;
+};
+
+struct snd_enc_real32 {
+	u32 quant_bits;
+	u32 start_region;
+	u32 num_regions;
+};
+
+struct snd_enc_flac32 {
+	u32 num;
+	u32 gain;
+};
+
+struct snd_enc_generic32 {
+	u32 bw;	/* encoder bandwidth */
+	s32 reserved[15];
+};
+struct snd_dec_dts32 {
+	u32 modelIdLength;
+	compat_uptr_t modelId;
+};
+struct snd_dec_ddp32 {
+	u32 params_length;
+	compat_uptr_t params;
+	u32 params_id[18];
+	u32 params_value[18];
+};
+
+union snd_codec_options32 {
+	struct snd_enc_wma32 wma;
+	struct snd_enc_vorbis32 vorbis;
+	struct snd_enc_real32 real;
+	struct snd_enc_flac32 flac;
+	struct snd_enc_generic32 generic;
+	struct snd_dec_dts32 dts;
+	struct snd_dec_ddp32 ddp;
+};
+
+struct snd_codec32 {
+	u32 id;
+	u32 ch_in;
+	u32 ch_out;
+	u32 sample_rate;
+	u32 bit_rate;
+	u32 rate_control;
+	u32 profile;
+	u32 level;
+	u32 ch_mode;
+	u32 format;
+	u32 align;
+	u32 transcode_dts;
+	struct snd_dec_dts32 dts;
+	union snd_codec_options32 options;
+	u32 reserved[3];
+};
+
+struct snd_compressed_buffer32 {
+	u32 fragment_size;
+	u32 fragments;
+};
+
+struct snd_compr_params32 {
+	struct snd_compressed_buffer32 buffer;
+	struct snd_codec32 codec;
+	u8 no_wake_mode;
+};
+
+struct snd_compr_caps32 {
+	u32 num_codecs;
+	u32 direction;
+	u32 min_fragment_size;
+	u32 max_fragment_size;
+	u32 min_fragments;
+	u32 max_fragments;
+	u32 codecs[MAX_NUM_CODECS];
+	u32 reserved[11];
+};
+struct snd_compr_tstamp32 {
+	u32 byte_offset;
+	u32 copied_total;
+	compat_ulong_t pcm_frames;
+	compat_ulong_t pcm_io_frames;
+	u32 sampling_rate;
+	compat_u64 timestamp;
+};
+enum {
+	SNDRV_COMPRESS_TSTAMP32 = _IOR('C', 0x20, struct snd_compr_tstamp32),
+	SNDRV_COMPRESS_GET_CAPS32 = _IOWR('C', 0x10, struct snd_compr_caps32),
+	SNDRV_COMPRESS_SET_PARAMS32 =
+	_IOW('C', 0x12, struct snd_compr_params32),
+};
+static int msm_compr_compat_ioctl(struct snd_pcm_substream *substream,
+		unsigned int cmd, void *arg)
+{
+	int err = 0;
+	switch (cmd) {
+	case SNDRV_COMPRESS_TSTAMP32: {
+		struct snd_compr_tstamp tstamp;
+		struct snd_compr_tstamp32 tstamp32;
+		memset(&tstamp, 0, sizeof(tstamp));
+		memset(&tstamp32, 0, sizeof(tstamp32));
+		cmd = SNDRV_COMPRESS_TSTAMP;
+		err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
+		if (err) {
+			pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
+			__func__, err);
+			goto bail_out;
+		}
+		tstamp32.byte_offset = tstamp.byte_offset;
+		tstamp32.copied_total = tstamp.copied_total;
+		tstamp32.pcm_frames = tstamp.pcm_frames;
+		tstamp32.pcm_io_frames = tstamp.pcm_io_frames;
+		tstamp32.sampling_rate = tstamp.sampling_rate;
+		tstamp32.timestamp = tstamp.timestamp;
+		if (copy_to_user(arg, &tstamp32, sizeof(tstamp32))) {
+			pr_err("%s: copytouser failed COMPRESS_TSTAMP32\n",
+			__func__);
+			err = -EFAULT;
+		}
+		break;
+	}
+	case SNDRV_COMPRESS_GET_CAPS32: {
+		struct snd_compr_caps caps;
+		struct snd_compr_caps32 caps32;
+		u32 i;
+		memset(&caps, 0, sizeof(caps));
+		memset(&caps32, 0, sizeof(caps32));
+		cmd = SNDRV_COMPRESS_GET_CAPS;
+		err = msm_compr_ioctl_shared(substream, cmd, &caps);
+		if (err) {
+			pr_err("%s: GET_CAPS failed rc %d\n",
+			__func__, err);
+			goto bail_out;
+		}
+		pr_debug("SNDRV_COMPRESS_GET_CAPS_32\n");
+		if (!err && caps.num_codecs >= MAX_NUM_CODECS) {
+			pr_err("%s: Invalid number of codecs\n", __func__);
+			err = -EINVAL;
+			goto bail_out;
+		}
+		caps32.direction = caps.direction;
+		caps32.max_fragment_size = caps.max_fragment_size;
+		caps32.max_fragments = caps.max_fragments;
+		caps32.min_fragment_size = caps.min_fragment_size;
+		caps32.num_codecs = caps.num_codecs;
+		for (i = 0; i < caps.num_codecs; i++)
+			caps32.codecs[i] = caps.codecs[i];
+		if (copy_to_user(arg, &caps32, sizeof(caps32))) {
+			pr_err("%s: copytouser failed COMPRESS_GETCAPS32\n",
+			__func__);
+			err = -EFAULT;
+		}
+		break;
+	}
+	case SNDRV_COMPRESS_SET_PARAMS32: {
+		struct snd_compr_params32 params32;
+		struct snd_compr_params params;
+		memset(&params32, 0 , sizeof(params32));
+		memset(&params, 0 , sizeof(params));
+		cmd = SNDRV_COMPRESS_SET_PARAMS;
+		if (copy_from_user(&params32, arg, sizeof(params32))) {
+			pr_err("%s: copyfromuser failed SET_PARAMS32\n",
+			__func__);
+			err = -EFAULT;
+			goto bail_out;
+		}
+		params.no_wake_mode = params32.no_wake_mode;
+		params.codec.id = params32.codec.id;
+		params.codec.ch_in = params32.codec.ch_in;
+		params.codec.ch_out = params32.codec.ch_out;
+		params.codec.sample_rate = params32.codec.sample_rate;
+		params.codec.bit_rate = params32.codec.bit_rate;
+		params.codec.rate_control = params32.codec.rate_control;
+		params.codec.profile = params32.codec.profile;
+		params.codec.level = params32.codec.level;
+		params.codec.ch_mode = params32.codec.ch_mode;
+		params.codec.format = params32.codec.format;
+		params.codec.align = params32.codec.align;
+		params.codec.transcode_dts = params32.codec.transcode_dts;
+
+		switch (params.codec.id) {
+		case SND_AUDIOCODEC_WMA:
+		case SND_AUDIOCODEC_WMA_PRO:
+			params.codec.options.wma.encodeopt1 =
+			params32.codec.options.wma.encodeopt1;
+			params.codec.options.wma.encodeopt2 =
+			params32.codec.options.wma.encodeopt2;
+			params.codec.options.wma.super_block_align =
+			params32.codec.options.wma.super_block_align;
+		break;
+		case SND_AUDIOCODEC_VORBIS:
+			params.codec.options.vorbis.downmix =
+			params32.codec.options.vorbis.downmix;
+			params.codec.options.vorbis.managed =
+			params32.codec.options.vorbis.managed;
+			params.codec.options.vorbis.max_bit_rate =
+			params32.codec.options.vorbis.max_bit_rate;
+			params.codec.options.vorbis.min_bit_rate =
+			params32.codec.options.vorbis.min_bit_rate;
+			params.codec.options.vorbis.quality =
+			params32.codec.options.vorbis.quality;
+		break;
+		case SND_AUDIOCODEC_REAL:
+			params.codec.options.real.num_regions =
+			params32.codec.options.real.num_regions;
+			params.codec.options.real.quant_bits =
+			params32.codec.options.real.quant_bits;
+			params.codec.options.real.start_region =
+			params32.codec.options.real.start_region;
+		break;
+		case SND_AUDIOCODEC_FLAC:
+			params.codec.options.flac.gain =
+			params32.codec.options.flac.gain;
+			params.codec.options.flac.num =
+			params32.codec.options.flac.num;
+		break;
+		case SND_AUDIOCODEC_DTS:
+		case SND_AUDIOCODEC_DTS_PASS_THROUGH:
+		case SND_AUDIOCODEC_DTS_LBR:
+		case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
+		case SND_AUDIOCODEC_DTS_TRANSCODE_LOOPBACK:
+			params.codec.options.dts.modelIdLength =
+			params32.codec.options.dts.modelIdLength;
+			params.codec.options.dts.modelId =
+			compat_ptr(params32.codec.options.dts.modelId);
+		break;
+		case SND_AUDIOCODEC_AC3:
+		case SND_AUDIOCODEC_EAC3:
+			params.codec.options.ddp.params_length =
+			params32.codec.options.ddp.params_length;
+			params.codec.options.ddp.params =
+			compat_ptr(params32.codec.options.ddp.params);
+			memcpy(params.codec.options.ddp.params_value,
+			params32.codec.options.ddp.params_value,
+			sizeof(params32.codec.options.ddp.params_value));
+			memcpy(params.codec.options.ddp.params_id,
+			params32.codec.options.ddp.params_id,
+			sizeof(params32.codec.options.ddp.params_id));
+		break;
+		default:
+			params.codec.options.generic.bw =
+			params32.codec.options.generic.bw;
+		break;
+		}
+		if (!err)
+			err = msm_compr_ioctl_shared(substream, cmd, &params);
+		break;
+	}
+	default:
+		err = msm_compr_ioctl_shared(substream, cmd, arg);
+	}
+bail_out:
+	return err;
+
+}
+#endif
+static int msm_compr_ioctl(struct snd_pcm_substream *substream,
+		unsigned int cmd, void *arg)
+{
+	int err = 0;
+	if (!substream) {
+		pr_err("%s: Invalid params\n", __func__);
+		return -EINVAL;
+	}
+	pr_debug("%s called with cmd = %d\n", __func__, cmd);
+	switch (cmd) {
+	case SNDRV_COMPRESS_TSTAMP: {
+		struct snd_compr_tstamp tstamp;
+		if (!arg) {
+			pr_err("%s: Invalid params Tstamp\n", __func__);
+			return -EINVAL;
+		}
+		err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
+		if (err)
+			pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
+			__func__, err);
+		if (!err && copy_to_user(arg, &tstamp, sizeof(tstamp))) {
+			pr_err("%s: copytouser failed COMPRESS_TSTAMP\n",
+			__func__);
+			err = -EFAULT;
+		}
+		break;
+	}
+	case SNDRV_COMPRESS_GET_CAPS: {
+		struct snd_compr_caps cap;
+		if (!arg) {
+			pr_err("%s: Invalid params getcaps\n", __func__);
+			return -EINVAL;
+		}
+		pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
+		err = msm_compr_ioctl_shared(substream, cmd, &cap);
+		if (err)
+			pr_err("%s: GET_CAPS failed rc %d\n",
+			__func__, err);
+		if (!err && copy_to_user(arg, &cap, sizeof(cap))) {
+			pr_err("%s: copytouser failed GET_CAPS\n",
+			__func__);
+			err = -EFAULT;
+		}
+		break;
+	}
+	case SNDRV_COMPRESS_SET_PARAMS: {
+		struct snd_compr_params params;
+		if (!arg) {
+			pr_err("%s: Invalid params setparam\n", __func__);
+			return -EINVAL;
+		}
+		if (copy_from_user(&params, arg,
+			sizeof(struct snd_compr_params))) {
+			pr_err("%s: SET_PARAMS\n", __func__);
+			return -EFAULT;
+		}
+		err = msm_compr_ioctl_shared(substream, cmd, &params);
+		if (err)
+			pr_err("%s: SET_PARAMS failed rc %d\n",
+			__func__, err);
+		break;
+	}
+	default:
+		err = msm_compr_ioctl_shared(substream, cmd, arg);
+	}
+	return err;
+}
+
+static int msm_compr_restart(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct compr_audio *compr = runtime->private_data;
+	struct msm_audio *prtd = &compr->prtd;
+	struct audio_aio_write_param param;
+	struct audio_buffer *buf = NULL;
+	struct output_meta_data_st output_meta_data;
+	int time_stamp_flag = 0;
+	int buffer_length = 0;
+
+	pr_debug("%s, trigger restart\n", __func__);
+
+	if (runtime->render_flag & SNDRV_RENDER_STOPPED) {
+		buf = prtd->audio_client->port[IN].buf;
+		pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
+				__func__, prtd->pcm_count, prtd->out_head);
+		pr_debug("%s:writing buffer[%d] from 0x%08x\n",
+				__func__, prtd->out_head,
+				((unsigned int)buf[0].phys
+				+ (prtd->out_head * prtd->pcm_count)));
+
+		if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+			time_stamp_flag = SET_TIMESTAMP;
+		else
+			time_stamp_flag = NO_TIMESTAMP;
+		memcpy(&output_meta_data, (char *)(buf->data +
+			prtd->out_head * prtd->pcm_count),
+			COMPRE_OUTPUT_METADATA_SIZE);
+
+		buffer_length = output_meta_data.frame_size;
+		pr_debug("meta_data_length: %d, frame_length: %d\n",
+			 output_meta_data.meta_data_length,
+			 output_meta_data.frame_size);
+		pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
+			 output_meta_data.timestamp_msw,
+			 output_meta_data.timestamp_lsw);
+
+		param.paddr = (unsigned long)buf[0].phys
+				+ (prtd->out_head * prtd->pcm_count)
+				+ output_meta_data.meta_data_length;
+		param.len = buffer_length;
+		param.msw_ts = output_meta_data.timestamp_msw;
+		param.lsw_ts = output_meta_data.timestamp_lsw;
+		param.flags = time_stamp_flag;
+		param.uid = prtd->session_id;
+		if (q6asm_async_write(prtd->audio_client,
+					&param) < 0)
+			pr_err("%s:q6asm_async_write failed\n",
+				__func__);
+		else
+			prtd->out_head =
+				(prtd->out_head + 1) & (runtime->periods - 1);
+
+		runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
+		return 0;
+	}
+	return 0;
+}
+
+static int msm_compr_volume_ctl_put(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	int rc = 0;
+	struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
+	struct snd_pcm_substream *substream =
+			 vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	struct msm_audio *prtd;
+	int volume = ucontrol->value.integer.value[0];
+
+	pr_debug("%s: volume : %x\n", __func__, volume);
+	if (!substream)
+		return -ENODEV;
+	if (!substream->runtime)
+		return 0;
+	prtd = substream->runtime->private_data;
+	if (prtd)
+		rc = compressed_set_volume(prtd, volume);
+
+	return rc;
+}
+
+static int msm_compr_volume_ctl_get(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
+	struct snd_pcm_substream *substream =
+			 vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	struct msm_audio *prtd;
+
+	pr_debug("%s\n", __func__);
+	if (!substream)
+		return -ENODEV;
+	if (!substream->runtime)
+		return 0;
+	prtd = substream->runtime->private_data;
+	if (prtd)
+		ucontrol->value.integer.value[0] = prtd->volume;
+	return 0;
+}
+
+static int msm_compr_add_controls(struct snd_soc_pcm_runtime *rtd)
+{
+	int ret = 0;
+	struct snd_pcm *pcm = rtd->pcm;
+	struct snd_pcm_volume *volume_info;
+	struct snd_kcontrol *kctl;
+
+	dev_dbg(rtd->dev, "%s, Volume cntrl add\n", __func__);
+	ret = snd_pcm_add_volume_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+				      NULL, 1, rtd->dai_link->be_id,
+				      &volume_info);
+	if (ret < 0)
+		return ret;
+	kctl = volume_info->kctl;
+	kctl->put = msm_compr_volume_ctl_put;
+	kctl->get = msm_compr_volume_ctl_get;
+	kctl->tlv.p = compr_rx_vol_gain;
+	return 0;
+}
+
+static struct snd_pcm_ops msm_compr_ops = {
+	.open	   = msm_compr_open,
+	.hw_params	= msm_compr_hw_params,
+	.close	  = msm_compr_close,
+	.ioctl	  = msm_compr_ioctl,
+	.prepare	= msm_compr_prepare,
+	.trigger	= msm_compr_trigger,
+	.pointer	= msm_compr_pointer,
+	.mmap		= msm_compr_mmap,
+	.restart	= msm_compr_restart,
+#ifdef CONFIG_COMPAT
+	.compat_ioctl   = msm_compr_compat_ioctl,
+#endif
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	int ret = 0;
+
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+	ret = msm_compr_add_controls(rtd);
+	if (ret)
+		pr_err("%s, kctl add failed\n", __func__);
+	return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+	.ops		= &msm_compr_ops,
+	.pcm_new	= msm_asoc_pcm_new,
+};
+
+static int msm_compr_probe(struct platform_device *pdev)
+{
+	if (pdev->dev.of_node)
+		dev_set_name(&pdev->dev, "%s", "msm-compr-dsp");
+
+	dev_info(&pdev->dev, "%s: dev name %s\n",
+			 __func__, dev_name(&pdev->dev));
+
+	atomic_set(&compressed_audio.audio_ocmem_req, 0);
+	return snd_soc_register_platform(&pdev->dev,
+				   &msm_soc_platform);
+}
+
+static int msm_compr_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static const struct of_device_id msm_compr_dt_match[] = {
+	{.compatible = "qcom,msm-compr-dsp"},
+	{}
+};
+MODULE_DEVICE_TABLE(of, msm_compr_dt_match);
+
+static struct platform_driver msm_compr_driver = {
+	.driver = {
+		.name = "msm-compr-dsp",
+		.owner = THIS_MODULE,
+		.of_match_table = msm_compr_dt_match,
+	},
+	.probe = msm_compr_probe,
+	.remove = msm_compr_remove,
+};
+
+static int __init msm_soc_platform_init(void)
+{
+	init_waitqueue_head(&the_locks.enable_wait);
+	init_waitqueue_head(&the_locks.eos_wait);
+	init_waitqueue_head(&the_locks.write_wait);
+	init_waitqueue_head(&the_locks.read_wait);
+	init_waitqueue_head(&the_locks.flush_wait);
+
+	return platform_driver_register(&msm_compr_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+	platform_driver_unregister(&msm_compr_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
new file mode 100644
index 000000000000..d6e3ec6956b1
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2012, The Linux Foundation. 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 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef _MSM_COMPR_H
+#define _MSM_COMPR_H
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+#include <sound/compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+
+#include "msm-pcm-q6-v2.h"
+
+struct compr_info {
+	struct snd_compr_caps compr_cap;
+	struct snd_compr_codec_caps codec_caps;
+	struct snd_compr_params codec_param;
+};
+
+struct compr_audio {
+	struct msm_audio prtd;
+	struct compr_info info;
+	uint32_t codec;
+};
+
+#endif /*_MSM_COMPR_H*/